renamed src/audioOutputs/ to src/output/

Again, no CamelCase in the directory name.

13 files changed, 0 insertions, 4179 deletions

diff --git a/src/audioOutputs/audioOutput_alsa.c b/src/audioOutputs/audioOutput_alsa.c
deleted file mode 100644
index 1845f1b76..000000000
--- a/src/audioOutputs/audioOutput_alsa.c
+++ /dev/null
@@ -1,444 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_ALSA
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-static const char default_device[] = "default";
-
-#define MPD_ALSA_RETRY_NR 5
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <alsa/asoundlib.h>
-
-typedef snd_pcm_sframes_t alsa_writei_t(snd_pcm_t * pcm, const void *buffer,
-					snd_pcm_uframes_t size);
-
-typedef struct _AlsaData {
-	const char *device;
-
-	/** the mode flags passed to snd_pcm_open */
-	int mode;
-
-	snd_pcm_t *pcmHandle;
-	alsa_writei_t *writei;
-	unsigned int buffer_time;
-	unsigned int period_time;
-	int sampleSize;
-	int useMmap;
-} AlsaData;
-
-static AlsaData *newAlsaData(void)
-{
-	AlsaData *ret = xmalloc(sizeof(AlsaData));
-
-	ret->device = default_device;
-	ret->mode = 0;
-	ret->pcmHandle = NULL;
-	ret->writei = snd_pcm_writei;
-	ret->useMmap = 0;
-	ret->buffer_time = 0;
-	ret->period_time = 0;
-
-	return ret;
-}
-
-static void freeAlsaData(AlsaData * ad)
-{
-	if (ad->device && ad->device != default_device)
-		xfree(ad->device);
-	free(ad);
-}
-
-static void
-alsa_configure(AlsaData *ad, ConfigParam *param)
-{
-	BlockParam *bp;
-
-	if ((bp = getBlockParam(param, "device")))
-		ad->device = xstrdup(bp->value);
-	ad->useMmap = getBoolBlockParam(param, "use_mmap", 1);
-	if (ad->useMmap == CONF_BOOL_UNSET)
-		ad->useMmap = 0;
-	if ((bp = getBlockParam(param, "buffer_time")))
-		ad->buffer_time = atoi(bp->value);
-	if ((bp = getBlockParam(param, "period_time")))
-		ad->period_time = atoi(bp->value);
-
-#ifdef SND_PCM_NO_AUTO_RESAMPLE
-	if (!getBoolBlockParam(param, "auto_resample", true))
-		ad->mode |= SND_PCM_NO_AUTO_RESAMPLE;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_CHANNELS
-	if (!getBoolBlockParam(param, "auto_channels", true))
-		ad->mode |= SND_PCM_NO_AUTO_CHANNELS;
-#endif
-
-#ifdef SND_PCM_NO_AUTO_FORMAT
-	if (!getBoolBlockParam(param, "auto_format", true))
-		ad->mode |= SND_PCM_NO_AUTO_FORMAT;
-#endif
-}
-
-static void *alsa_initDriver(mpd_unused struct audio_output *ao,
-			     mpd_unused const struct audio_format *audio_format,
-			     ConfigParam * param)
-{
-	/* no need for pthread_once thread-safety when reading config */
-	static int free_global_registered;
-	AlsaData *ad = newAlsaData();
-
-	if (!free_global_registered) {
-		atexit((void(*)(void))snd_config_update_free_global);
-		free_global_registered = 1;
-	}
-
-	if (param)
-		alsa_configure(ad, param);
-
-	return ad;
-}
-
-static void alsa_finishDriver(void *data)
-{
-	AlsaData *ad = data;
-
-	freeAlsaData(ad);
-}
-
-static int alsa_testDefault(void)
-{
-	snd_pcm_t *handle;
-
-	int ret = snd_pcm_open(&handle, default_device,
-	                       SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
-	if (ret) {
-		WARNING("Error opening default ALSA device: %s\n",
-			snd_strerror(-ret));
-		return -1;
-	} else
-		snd_pcm_close(handle);
-
-	return 0;
-}
-
-static snd_pcm_format_t get_bitformat(const struct audio_format *af)
-{
-	switch (af->bits) {
-	case 8: return SND_PCM_FORMAT_S8;
-	case 16: return SND_PCM_FORMAT_S16;
-	case 24: return SND_PCM_FORMAT_S24;
-	case 32: return SND_PCM_FORMAT_S32;
-	}
-	return SND_PCM_FORMAT_UNKNOWN;
-}
-
-static int alsa_openDevice(void *data, struct audio_format *audioFormat)
-{
-	AlsaData *ad = data;
-	snd_pcm_format_t bitformat;
-	snd_pcm_hw_params_t *hwparams;
-	snd_pcm_sw_params_t *swparams;
-	unsigned int sample_rate = audioFormat->sample_rate;
-	unsigned int channels = audioFormat->channels;
-	snd_pcm_uframes_t alsa_buffer_size;
-	snd_pcm_uframes_t alsa_period_size;
-	int err;
-	const char *cmd = NULL;
-	int retry = MPD_ALSA_RETRY_NR;
-	unsigned int period_time, period_time_ro;
-	unsigned int buffer_time;
-
-	if ((bitformat = get_bitformat(audioFormat)) == SND_PCM_FORMAT_UNKNOWN)
-		ERROR("ALSA device \"%s\" doesn't support %u bit audio\n",
-		      ad->device, audioFormat->bits);
-
-	err = snd_pcm_open(&ad->pcmHandle, ad->device,
-			   SND_PCM_STREAM_PLAYBACK, ad->mode);
-	if (err < 0) {
-		ad->pcmHandle = NULL;
-		goto error;
-	}
-
-	period_time_ro = period_time = ad->period_time;
-configure_hw:
-	/* configure HW params */
-	snd_pcm_hw_params_alloca(&hwparams);
-
-	cmd = "snd_pcm_hw_params_any";
-	err = snd_pcm_hw_params_any(ad->pcmHandle, hwparams);
-	if (err < 0)
-		goto error;
-
-	if (ad->useMmap) {
-		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
-						   SND_PCM_ACCESS_MMAP_INTERLEAVED);
-		if (err < 0) {
-			ERROR("Cannot set mmap'ed mode on ALSA device \"%s\": "
-			      " %s\n", ad->device, snd_strerror(-err));
-			ERROR("Falling back to direct write mode\n");
-			ad->useMmap = 0;
-		} else
-			ad->writei = snd_pcm_mmap_writei;
-	}
-
-	if (!ad->useMmap) {
-		cmd = "snd_pcm_hw_params_set_access";
-		err = snd_pcm_hw_params_set_access(ad->pcmHandle, hwparams,
-						   SND_PCM_ACCESS_RW_INTERLEAVED);
-		if (err < 0)
-			goto error;
-		ad->writei = snd_pcm_writei;
-	}
-
-	err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams, bitformat);
-	if (err == -EINVAL && audioFormat->bits != 16) {
-		/* fall back to 16 bit, let pcm_utils.c do the conversion */
-		err = snd_pcm_hw_params_set_format(ad->pcmHandle, hwparams,
-						   SND_PCM_FORMAT_S16);
-		if (err == 0) {
-			DEBUG("ALSA device \"%s\": converting %u bit to 16 bit\n",
-			      ad->device, audioFormat->bits);
-			audioFormat->bits = 16;
-		}
-	}
-
-	if (err < 0) {
-		ERROR("ALSA device \"%s\" does not support %u bit audio: "
-		      "%s\n", ad->device, audioFormat->bits, snd_strerror(-err));
-		goto fail;
-	}
-
-	err = snd_pcm_hw_params_set_channels_near(ad->pcmHandle, hwparams,
-						  &channels);
-	if (err < 0) {
-		ERROR("ALSA device \"%s\" does not support %i channels: "
-		      "%s\n", ad->device, (int)audioFormat->channels,
-		      snd_strerror(-err));
-		goto fail;
-	}
-	audioFormat->channels = (int8_t)channels;
-
-	err = snd_pcm_hw_params_set_rate_near(ad->pcmHandle, hwparams,
-					      &sample_rate, NULL);
-	if (err < 0 || sample_rate == 0) {
-		ERROR("ALSA device \"%s\" does not support %u Hz audio\n",
-		      ad->device, audioFormat->sample_rate);
-		goto fail;
-	}
-	audioFormat->sample_rate = sample_rate;
-
-	if (ad->buffer_time > 0) {
-		buffer_time = ad->buffer_time;
-		cmd = "snd_pcm_hw_params_set_buffer_time_near";
-		err = snd_pcm_hw_params_set_buffer_time_near(ad->pcmHandle, hwparams,
-							     &buffer_time, NULL);
-		if (err < 0)
-			goto error;
-	}
-
-	if (period_time_ro > 0) {
-		period_time = period_time_ro;
-		cmd = "snd_pcm_hw_params_set_period_time_near";
-		err = snd_pcm_hw_params_set_period_time_near(ad->pcmHandle, hwparams,
-							     &period_time, NULL);
-		if (err < 0)
-			goto error;
-	}
-
-	cmd = "snd_pcm_hw_params";
-	err = snd_pcm_hw_params(ad->pcmHandle, hwparams);
-	if (err == -EPIPE && --retry > 0 && period_time_ro > 0) {
-		period_time_ro = period_time_ro >> 1;
-		goto configure_hw;
-	} else if (err < 0)
-		goto error;
-	if (retry != MPD_ALSA_RETRY_NR)
-		DEBUG("ALSA period_time set to %d\n", period_time);
-
-	cmd = "snd_pcm_hw_params_get_buffer_size";
-	err = snd_pcm_hw_params_get_buffer_size(hwparams, &alsa_buffer_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_hw_params_get_period_size";
-	err = snd_pcm_hw_params_get_period_size(hwparams, &alsa_period_size,
-						NULL);
-	if (err < 0)
-		goto error;
-
-	/* configure SW params */
-	snd_pcm_sw_params_alloca(&swparams);
-
-	cmd = "snd_pcm_sw_params_current";
-	err = snd_pcm_sw_params_current(ad->pcmHandle, swparams);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_start_threshold";
-	err = snd_pcm_sw_params_set_start_threshold(ad->pcmHandle, swparams,
-						    alsa_buffer_size -
-						    alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params_set_avail_min";
-	err = snd_pcm_sw_params_set_avail_min(ad->pcmHandle, swparams,
-					      alsa_period_size);
-	if (err < 0)
-		goto error;
-
-	cmd = "snd_pcm_sw_params";
-	err = snd_pcm_sw_params(ad->pcmHandle, swparams);
-	if (err < 0)
-		goto error;
-
-	ad->sampleSize = audio_format_frame_size(audioFormat);
-
-	DEBUG("ALSA device \"%s\" will be playing %i bit, %u channel audio at "
-	      "%u Hz\n", ad->device, audioFormat->bits,
-	      channels, sample_rate);
-
-	return 0;
-
-error:
-	if (cmd) {
-		ERROR("Error opening ALSA device \"%s\" (%s): %s\n",
-		      ad->device, cmd, snd_strerror(-err));
-	} else {
-		ERROR("Error opening ALSA device \"%s\": %s\n", ad->device,
-		      snd_strerror(-err));
-	}
-fail:
-	if (ad->pcmHandle)
-		snd_pcm_close(ad->pcmHandle);
-	ad->pcmHandle = NULL;
-	return -1;
-}
-
-static int alsa_errorRecovery(AlsaData * ad, int err)
-{
-	if (err == -EPIPE) {
-		DEBUG("Underrun on ALSA device \"%s\"\n", ad->device);
-	} else if (err == -ESTRPIPE) {
-		DEBUG("ALSA device \"%s\" was suspended\n", ad->device);
-	}
-
-	switch (snd_pcm_state(ad->pcmHandle)) {
-	case SND_PCM_STATE_PAUSED:
-		err = snd_pcm_pause(ad->pcmHandle, /* disable */ 0);
-		break;
-	case SND_PCM_STATE_SUSPENDED:
-		err = snd_pcm_resume(ad->pcmHandle);
-		if (err == -EAGAIN)
-			return 0;
-		/* fall-through to snd_pcm_prepare: */
-	case SND_PCM_STATE_SETUP:
-	case SND_PCM_STATE_XRUN:
-		err = snd_pcm_prepare(ad->pcmHandle);
-		break;
-	case SND_PCM_STATE_DISCONNECTED:
-		/* so alsa_closeDevice won't try to drain: */
-		snd_pcm_close(ad->pcmHandle);
-		ad->pcmHandle = NULL;
-		break;
-	/* this is no error, so just keep running */
-	case SND_PCM_STATE_RUNNING:
-		err = 0;
-		break;
-	default:
-		/* unknown state, do nothing */
-		break;
-	}
-
-	return err;
-}
-
-static void alsa_dropBufferedAudio(void *data)
-{
-	AlsaData *ad = data;
-
-	alsa_errorRecovery(ad, snd_pcm_drop(ad->pcmHandle));
-}
-
-static void alsa_closeDevice(void *data)
-{
-	AlsaData *ad = data;
-
-	if (ad->pcmHandle) {
-		if (snd_pcm_state(ad->pcmHandle) == SND_PCM_STATE_RUNNING) {
-			snd_pcm_drain(ad->pcmHandle);
-		}
-		snd_pcm_close(ad->pcmHandle);
-		ad->pcmHandle = NULL;
-	}
-}
-
-static int alsa_playAudio(void *data, const char *playChunk, size_t size)
-{
-	AlsaData *ad = data;
-	int ret;
-
-	size /= ad->sampleSize;
-
-	while (size > 0) {
-		ret = ad->writei(ad->pcmHandle, playChunk, size);
-
-		if (ret == -EAGAIN || ret == -EINTR)
-			continue;
-
-		if (ret < 0) {
-			if (alsa_errorRecovery(ad, ret) < 0) {
-				ERROR("closing ALSA device \"%s\" due to write "
-				      "error: %s\n", ad->device,
-				      snd_strerror(-errno));
-				alsa_closeDevice(ad);
-				return -1;
-			}
-			continue;
-		}
-
-		playChunk += ret * ad->sampleSize;
-		size -= ret;
-	}
-
-	return 0;
-}
-
-const struct audio_output_plugin alsaPlugin = {
-	.name = "alsa",
-	.test_default_device = alsa_testDefault,
-	.init = alsa_initDriver,
-	.finish = alsa_finishDriver,
-	.open = alsa_openDevice,
-	.play = alsa_playAudio,
-	.cancel = alsa_dropBufferedAudio,
-	.close = alsa_closeDevice,
-};
-
-#else /* HAVE ALSA */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(alsaPlugin)
-#endif /* HAVE_ALSA */
diff --git a/src/audioOutputs/audioOutput_ao.c b/src/audioOutputs/audioOutput_ao.c
deleted file mode 100644
index e731f972a..000000000
--- a/src/audioOutputs/audioOutput_ao.c
+++ /dev/null
@@ -1,253 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_AO
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <ao/ao.h>
-
-static int driverInitCount;
-
-typedef struct _AoData {
-	int writeSize;
-	int driverId;
-	ao_option *options;
-	ao_device *device;
-} AoData;
-
-static AoData *newAoData(void)
-{
-	AoData *ret = xmalloc(sizeof(AoData));
-	ret->device = NULL;
-	ret->options = NULL;
-
-	return ret;
-}
-
-static void audioOutputAo_error(void)
-{
-	if (errno == AO_ENOTLIVE) {
-		ERROR("not a live ao device\n");
-	} else if (errno == AO_EOPENDEVICE) {
-		ERROR("not able to open audio device\n");
-	} else if (errno == AO_EBADOPTION) {
-		ERROR("bad driver option\n");
-	}
-}
-
-static void *audioOutputAo_initDriver(struct audio_output *ao,
-				      mpd_unused const struct audio_format *audio_format,
-				      ConfigParam * param)
-{
-	ao_info *ai;
-	char *duplicated;
-	char *stk1;
-	char *stk2;
-	char *n1;
-	char *key;
-	char *value;
-	char *test;
-	AoData *ad = newAoData();
-	BlockParam *blockParam;
-
-	if ((blockParam = getBlockParam(param, "write_size"))) {
-		ad->writeSize = strtol(blockParam->value, &test, 10);
-		if (*test != '\0') {
-			FATAL("\"%s\" is not a valid write size at line %i\n",
-			      blockParam->value, blockParam->line);
-		}
-	} else
-		ad->writeSize = 1024;
-
-	if (driverInitCount == 0) {
-		ao_initialize();
-	}
-	driverInitCount++;
-
-	blockParam = getBlockParam(param, "driver");
-
-	if (!blockParam || 0 == strcmp(blockParam->value, "default")) {
-		ad->driverId = ao_default_driver_id();
-	} else if ((ad->driverId = ao_driver_id(blockParam->value)) < 0) {
-		FATAL("\"%s\" is not a valid ao driver at line %i\n",
-		      blockParam->value, blockParam->line);
-	}
-
-	if ((ai = ao_driver_info(ad->driverId)) == NULL) {
-		FATAL("problems getting driver info for device defined at line %i\n"
-		      "you may not have permission to the audio device\n", param->line);
-	}
-
-	DEBUG("using ao driver \"%s\" for \"%s\"\n", ai->short_name,
-	      audio_output_get_name(ao));
-
-	blockParam = getBlockParam(param, "options");
-
-	if (blockParam) {
-		duplicated = xstrdup(blockParam->value);
-	} else
-		duplicated = xstrdup("");
-
-	if (strlen(duplicated)) {
-		stk1 = NULL;
-		n1 = strtok_r(duplicated, ";", &stk1);
-		while (n1) {
-			stk2 = NULL;
-			key = strtok_r(n1, "=", &stk2);
-			if (!key)
-				FATAL("problems parsing options \"%s\"\n", n1);
-			/*found = 0;
-			   for(i=0;i<ai->option_count;i++) {
-			   if(strcmp(ai->options[i],key)==0) {
-			   found = 1;
-			   break;
-			   }
-			   }
-			   if(!found) {
-			   FATAL("\"%s\" is not an option for "
-			   "\"%s\" ao driver\n",key,
-			   ai->short_name);
-			   } */
-			value = strtok_r(NULL, "", &stk2);
-			if (!value)
-				FATAL("problems parsing options \"%s\"\n", n1);
-			ao_append_option(&ad->options, key, value);
-			n1 = strtok_r(NULL, ";", &stk1);
-		}
-	}
-	free(duplicated);
-
-	return ad;
-}
-
-static void freeAoData(AoData * ad)
-{
-	ao_free_options(ad->options);
-	free(ad);
-}
-
-static void audioOutputAo_finishDriver(void *data)
-{
-	AoData *ad = (AoData *)data;
-	freeAoData(ad);
-
-	driverInitCount--;
-
-	if (driverInitCount == 0)
-		ao_shutdown();
-}
-
-static void audioOutputAo_dropBufferedAudio(mpd_unused void *data)
-{
-	/* not supported by libao */
-}
-
-static void audioOutputAo_closeDevice(void *data)
-{
-	AoData *ad = (AoData *)data;
-
-	if (ad->device) {
-		ao_close(ad->device);
-		ad->device = NULL;
-	}
-}
-
-static int audioOutputAo_openDevice(void *data,
-				    struct audio_format *audio_format)
-{
-	ao_sample_format format;
-	AoData *ad = (AoData *)data;
-
-	if (ad->device) {
-		audioOutputAo_closeDevice(ad);
-	}
-
-	format.bits = audio_format->bits;
-	format.rate = audio_format->sample_rate;
-	format.byte_format = AO_FMT_NATIVE;
-	format.channels = audio_format->channels;
-
-	ad->device = ao_open_live(ad->driverId, &format, ad->options);
-
-	if (ad->device == NULL)
-		return -1;
-
-	return 0;
-}
-
-/**
- * For whatever reason, libao wants a non-const pointer.  Let's hope
- * it does not write to the buffer, and use the union deconst hack to
- * work around this API misdesign.
- */
-static int ao_play_deconst(ao_device *device, const void *output_samples,
-			   uint_32 num_bytes)
-{
-	union {
-		const void *in;
-		void *out;
-	} u;
-
-	u.in = output_samples;
-	return ao_play(device, u.out, num_bytes);
-}
-
-static int audioOutputAo_play(void *data, const char *playChunk, size_t size)
-{
-	AoData *ad = (AoData *)data;
-	size_t chunk_size;
-
-	if (ad->device == NULL)
-		return -1;
-
-	while (size > 0) {
-		chunk_size = (size_t)ad->writeSize > size
-			? size : (size_t)ad->writeSize;
-
-		if (ao_play_deconst(ad->device, playChunk, chunk_size) == 0) {
-			audioOutputAo_error();
-			ERROR("closing audio device due to write error\n");
-			audioOutputAo_closeDevice(ad);
-			return -1;
-		}
-
-		playChunk += chunk_size;
-		size -= chunk_size;
-	}
-
-	return 0;
-}
-
-const struct audio_output_plugin aoPlugin = {
-	.name = "ao",
-	.init = audioOutputAo_initDriver,
-	.finish = audioOutputAo_finishDriver,
-	.open = audioOutputAo_openDevice,
-	.play = audioOutputAo_play,
-	.cancel = audioOutputAo_dropBufferedAudio,
-	.close = audioOutputAo_closeDevice,
-};
-
-#else
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(aoPlugin)
-#endif
diff --git a/src/audioOutputs/audioOutput_fifo.c b/src/audioOutputs/audioOutput_fifo.c
deleted file mode 100644
index d7eb91ff6..000000000
--- a/src/audioOutputs/audioOutput_fifo.c
+++ /dev/null
@@ -1,290 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_FIFO
-
-#include "../log.h"
-#include "../utils.h"
-#include "../timer.h"
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-
-#define FIFO_BUFFER_SIZE 65536 /* pipe capacity on Linux >= 2.6.11 */
-
-typedef struct _FifoData {
-	char *path;
-	int input;
-	int output;
-	int created;
-	Timer *timer;
-} FifoData;
-
-static FifoData *newFifoData(void)
-{
-	FifoData *ret;
-
-	ret = xmalloc(sizeof(FifoData));
-
-	ret->path = NULL;
-	ret->input = -1;
-	ret->output = -1;
-	ret->created = 0;
-	ret->timer = NULL;
-	
-	return ret;
-}
-
-static void freeFifoData(FifoData *fd)
-{
-	if (fd->path)
-		free(fd->path);
-
-	if (fd->timer)
-		timer_free(fd->timer);
-
-	free(fd);
-}
-
-static void removeFifo(FifoData *fd)
-{
-	DEBUG("Removing FIFO \"%s\"\n", fd->path);
-
-	if (unlink(fd->path) < 0) {
-		ERROR("Could not remove FIFO \"%s\": %s\n",
-		      fd->path, strerror(errno));
-		return;
-	}
-
-	fd->created = 0;
-}
-
-static void closeFifo(FifoData *fd)
-{
-	struct stat st;
-
-	if (fd->input >= 0) {
-		close(fd->input);
-		fd->input = -1;
-	}
-
-	if (fd->output >= 0) {
-		close(fd->output);
-		fd->output = -1;
-	}
-
-	if (fd->created && (stat(fd->path, &st) == 0))
-		removeFifo(fd);
-}
-
-static int makeFifo(FifoData *fd)
-{
-	if (mkfifo(fd->path, 0666) < 0) {
-		ERROR("Couldn't create FIFO \"%s\": %s\n",
-		      fd->path, strerror(errno));
-		return -1;
-	}
-
-	fd->created = 1;
-
-	return 0;
-}
-
-static int checkFifo(FifoData *fd)
-{
-	struct stat st;
-
-	if (stat(fd->path, &st) < 0) {
-		if (errno == ENOENT) {
-			/* Path doesn't exist */
-			return makeFifo(fd);
-		}
-
-		ERROR("Failed to stat FIFO \"%s\": %s\n",
-		      fd->path, strerror(errno));
-		return -1;
-	}
-
-	if (!S_ISFIFO(st.st_mode)) {
-		ERROR("\"%s\" already exists, but is not a FIFO\n", fd->path);
-		return -1;
-	}
-
-	return 0;
-}
-
-static int openFifo(FifoData *fd)
-{
-	if (checkFifo(fd) < 0)
-		return -1;
-
-	fd->input = open(fd->path, O_RDONLY|O_NONBLOCK);
-	if (fd->input < 0) {
-		ERROR("Could not open FIFO \"%s\" for reading: %s\n",
-		      fd->path, strerror(errno));
-		closeFifo(fd);
-		return -1;
-	}
-
-	fd->output = open(fd->path, O_WRONLY|O_NONBLOCK);
-	if (fd->output < 0) {
-		ERROR("Could not open FIFO \"%s\" for writing: %s\n",
-		      fd->path, strerror(errno));
-		closeFifo(fd);
-		return -1;
-	}
-
-	return 0;
-}
-
-static void *fifo_initDriver(mpd_unused struct audio_output *ao,
-			     mpd_unused const struct audio_format *audio_format,
-			     ConfigParam *param)
-{
-	FifoData *fd;
-	BlockParam *blockParam;
-	char *path;
-
-	blockParam = getBlockParam(param, "path");
-	if (!blockParam) {
-		FATAL("No \"path\" parameter specified for fifo output "
-		      "defined at line %i\n", param->line);
-	}
-
-	path = parsePath(blockParam->value);
-	if (!path) {
-		FATAL("Could not parse \"path\" parameter for fifo output "
-		      "at line %i\n", blockParam->line);
-	}
-
-	fd = newFifoData();
-	fd->path = path;
-
-	if (openFifo(fd) < 0) {
-		freeFifoData(fd);
-		return NULL;
-	}
-
-	return fd;
-}
-
-static void fifo_finishDriver(void *data)
-{
-	FifoData *fd = (FifoData *)data;
-
-	closeFifo(fd);
-	freeFifoData(fd);
-}
-
-static int fifo_openDevice(void *data,
-			   struct audio_format *audio_format)
-{
-	FifoData *fd = (FifoData *)data;
-
-	if (fd->timer)
-		timer_free(fd->timer);
-
-	fd->timer = timer_new(audio_format);
-
-	return 0;
-}
-
-static void fifo_closeDevice(void *data)
-{
-	FifoData *fd = (FifoData *)data;
-
-	if (fd->timer) {
-		timer_free(fd->timer);
-		fd->timer = NULL;
-	}
-}
-
-static void fifo_dropBufferedAudio(void *data)
-{
-	FifoData *fd = (FifoData *)data;
-	char buf[FIFO_BUFFER_SIZE];
-	int bytes = 1;
-
-	timer_reset(fd->timer);
-
-	while (bytes > 0 && errno != EINTR)
-		bytes = read(fd->input, buf, FIFO_BUFFER_SIZE);
-
-	if (bytes < 0 && errno != EAGAIN) {
-		WARNING("Flush of FIFO \"%s\" failed: %s\n",
-		        fd->path, strerror(errno));
-	}
-}
-
-static int fifo_playAudio(void *data,
-			  const char *playChunk, size_t size)
-{
-	FifoData *fd = (FifoData *)data;
-	size_t offset = 0;
-	ssize_t bytes;
-
-	if (!fd->timer->started)
-		timer_start(fd->timer);
-	else
-		timer_sync(fd->timer);
-
-	timer_add(fd->timer, size);
-
-	while (size) {
-		bytes = write(fd->output, playChunk + offset, size);
-		if (bytes < 0) {
-			switch (errno) {
-			case EAGAIN:
-				/* The pipe is full, so empty it */
-				fifo_dropBufferedAudio(fd);
-				continue;
-			case EINTR:
-				continue;
-			}
-
-			ERROR("Closing FIFO output \"%s\" due to write error: "
-			      "%s\n", fd->path, strerror(errno));
-			fifo_closeDevice(fd);
-			return -1;
-		}
-
-		size -= bytes;
-		offset += bytes;
-	}
-
-	return 0;
-}
-
-const struct audio_output_plugin fifoPlugin = {
-	.name = "fifo",
-	.init = fifo_initDriver,
-	.finish = fifo_finishDriver,
-	.open = fifo_openDevice,
-	.play = fifo_playAudio,
-	.cancel = fifo_dropBufferedAudio,
-	.close = fifo_closeDevice,
-};
-
-#else /* HAVE_FIFO */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(fifoPlugin)
-
-#endif /* !HAVE_FIFO */
diff --git a/src/audioOutputs/audioOutput_jack.c b/src/audioOutputs/audioOutput_jack.c
deleted file mode 100644
index 6f2bcd3a1..000000000
--- a/src/audioOutputs/audioOutput_jack.c
+++ /dev/null
@@ -1,486 +0,0 @@
-/* jack plug in for the Music Player Daemon (MPD)
- * (c)2006 by anarch(anarchsss@gmail.com)
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_JACK
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <assert.h>
-
-#include <jack/jack.h>
-#include <jack/types.h>
-#include <jack/ringbuffer.h>
-
-static const size_t sample_size = sizeof(jack_default_audio_sample_t);
-
-struct jack_data {
-	struct audio_output *ao;
-
-	/* configuration */
-	const char *name;
-	const char *output_ports[2];
-	int ringbuffer_size;
-
-	/* for srate() only */
-	struct audio_format *audio_format;
-
-	/* jack library stuff */
-	jack_port_t *ports[2];
-	jack_client_t *client;
-	jack_ringbuffer_t *ringbuffer[2];
-	int bps;
-	int shutdown;
-};
-
-static struct jack_data *
-mpd_jack_new(void)
-{
-	struct jack_data *ret;
-
-	ret = xcalloc(sizeof(*ret), 1);
-
-	ret->name = "mpd";
-	ret->ringbuffer_size = 32768;
-
-	return ret;
-}
-
-static void
-mpd_jack_client_free(struct jack_data *jd)
-{
-	assert(jd != NULL);
-
-	if (jd->client != NULL) {
-		jack_deactivate(jd->client);
-		jack_client_close(jd->client);
-		jd->client = NULL;
-	}
-
-	if (jd->ringbuffer[0] != NULL) {
-		jack_ringbuffer_free(jd->ringbuffer[0]);
-		jd->ringbuffer[0] = NULL;
-	}
-
-	if (jd->ringbuffer[1] != NULL) {
-		jack_ringbuffer_free(jd->ringbuffer[1]);
-		jd->ringbuffer[1] = NULL;
-	}
-}
-
-static void
-mpd_jack_free(struct jack_data *jd)
-{
-	int i;
-
-	assert(jd != NULL);
-
-	mpd_jack_client_free(jd);
-
-	if (strcmp(jd->name, "mpd") != 0)
-		xfree(jd->name);
-
-	for ( i = ARRAY_SIZE(jd->output_ports); --i >= 0; ) {
-		if (!jd->output_ports[i])
-			continue;
-		xfree(jd->output_ports[i]);
-	}
-
-	free(jd);
-}
-
-static void
-mpd_jack_finish(void *data)
-{
-	struct jack_data *jd = data;
-	mpd_jack_free(jd);
-}
-
-static int
-mpd_jack_srate(mpd_unused jack_nframes_t rate, void *data)
-{
-	struct jack_data *jd = (struct jack_data *)data;
-	struct audio_format *audioFormat = jd->audio_format;
-
-	audioFormat->sample_rate = (int)jack_get_sample_rate(jd->client);
-
-	return 0;
-}
-
-static int
-mpd_jack_process(jack_nframes_t nframes, void *arg)
-{
-	struct jack_data *jd = (struct jack_data *) arg;
-	jack_default_audio_sample_t *out;
-	size_t available;
-
-	if (nframes <= 0)
-		return 0;
-
-	for (unsigned i = 0; i < 2; ++i) {
-		available = jack_ringbuffer_read_space(jd->ringbuffer[i]);
-		assert(available % sample_size == 0);
-		available /= sample_size;
-		if (available > nframes)
-			available = nframes;
-
-		out = jack_port_get_buffer(jd->ports[i], nframes);
-		jack_ringbuffer_read(jd->ringbuffer[i],
-				     (char *)out, available * sample_size);
-
-		while (available < nframes)
-			/* ringbuffer underrun, fill with silence */
-			out[available++] = 0.0;
-	}
-
-	return 0;
-}
-
-static void
-mpd_jack_shutdown(void *arg)
-{
-	struct jack_data *jd = (struct jack_data *) arg;
-	jd->shutdown = 1;
-}
-
-static void
-set_audioformat(struct jack_data *jd, struct audio_format *audio_format)
-{
-	audio_format->sample_rate = jack_get_sample_rate(jd->client);
-	DEBUG("samplerate = %u\n", audio_format->sample_rate);
-	audio_format->channels = 2;
-
-	if (audio_format->bits != 16 && audio_format->bits != 24)
-		audio_format->bits = 24;
-
-	jd->bps = audio_format->channels
-		* sizeof(jack_default_audio_sample_t)
-		* audio_format->sample_rate;
-}
-
-static void
-mpd_jack_error(const char *msg)
-{
-	ERROR("jack: %s\n", msg);
-}
-
-static void *
-mpd_jack_init(struct audio_output *ao,
-	      mpd_unused const struct audio_format *audio_format,
-	      ConfigParam *param)
-{
-	struct jack_data *jd;
-	BlockParam *bp;
-	char *endptr;
-	int val;
-	char *cp = NULL;
-
-	jd = mpd_jack_new();
-	jd->ao = ao;
-
-	DEBUG("mpd_jack_init (pid=%d)\n", getpid());
-	if (param == NULL)
-		return jd;
-
-	if ( (bp = getBlockParam(param, "ports")) ) {
-		DEBUG("output_ports=%s\n", bp->value);
-
-		if (!(cp = strchr(bp->value, ',')))
-			FATAL("expected comma and a second value for '%s' "
-			      "at line %d: %s\n",
-			      bp->name, bp->line, bp->value);
-
-		*cp = '\0';
-		jd->output_ports[0] = xstrdup(bp->value);
-		*cp++ = ',';
-
-		if (!*cp)
-			FATAL("expected a second value for '%s' at line %d: "
-			      "%s\n", bp->name, bp->line, bp->value);
-
-		jd->output_ports[1] = xstrdup(cp);
-
-		if (strchr(cp,','))
-			FATAL("Only %d values are supported for '%s' "
-			      "at line %d\n",
-			      (int)ARRAY_SIZE(jd->output_ports),
-			      bp->name, bp->line);
-	}
-
-	if ( (bp = getBlockParam(param, "ringbuffer_size")) ) {
-		errno = 0;
-		val = strtol(bp->value, &endptr, 10);
-
-		if ( errno == 0 && endptr != bp->value) {
-			jd->ringbuffer_size = val < 32768 ? 32768 : val;
-			DEBUG("ringbuffer_size=%d\n", jd->ringbuffer_size);
-		} else {
-			FATAL("%s is not a number; ringbuf_size=%d\n",
-			      bp->value, jd->ringbuffer_size);
-		}
-	}
-
-	if ( (bp = getBlockParam(param, "name"))
-	     && (strcmp(bp->value, "mpd") != 0) ) {
-		jd->name = xstrdup(bp->value);
-		DEBUG("name=%s\n", jd->name);
-	}
-
-	return jd;
-}
-
-static int
-mpd_jack_test_default_device(void)
-{
-	return 0;
-}
-
-static int
-mpd_jack_connect(struct jack_data *jd, struct audio_format *audio_format)
-{
-	const char **jports;
-	char *port_name;
-
-	jd->audio_format = audio_format;
-
-	if ( (jd->client = jack_client_new(jd->name)) == NULL ) {
-		ERROR("jack server not running?\n");
-		return -1;
-	}
-
-	jack_set_error_function(mpd_jack_error);
-	jack_set_process_callback(jd->client, mpd_jack_process, jd);
-	jack_set_sample_rate_callback(jd->client, mpd_jack_srate, jd);
-	jack_on_shutdown(jd->client, mpd_jack_shutdown, jd);
-
-	if ( jack_activate(jd->client) ) {
-		ERROR("cannot activate client\n");
-		return -1;
-	}
-
-	jd->ports[0] = jack_port_register(jd->client, "left",
-					  JACK_DEFAULT_AUDIO_TYPE,
-					  JackPortIsOutput, 0);
-	if ( !jd->ports[0] ) {
-		ERROR("Cannot register left output port.\n");
-		return -1;
-	}
-
-	jd->ports[1] = jack_port_register(jd->client, "right",
-					  JACK_DEFAULT_AUDIO_TYPE,
-					  JackPortIsOutput, 0);
-	if ( !jd->ports[1] ) {
-		ERROR("Cannot register right output port.\n");
-		return -1;
-	}
-
-	/*  hay que buscar que hay  */
-	if (!jd->output_ports[1] &&
-	    (jports = jack_get_ports(jd->client, NULL, NULL,
-				     JackPortIsPhysical | JackPortIsInput))) {
-		jd->output_ports[0] = jports[0];
-		jd->output_ports[1] = jports[1] ? jports[1] : jports[0];
-		DEBUG("output_ports: %s %s\n",
-		      jd->output_ports[0], jd->output_ports[1]);
-		free(jports);
-	}
-
-	if ( jd->output_ports[1] ) {
-		jd->ringbuffer[0] = jack_ringbuffer_create(jd->ringbuffer_size);
-		jd->ringbuffer[1] = jack_ringbuffer_create(jd->ringbuffer_size);
-		memset(jd->ringbuffer[0]->buf, 0, jd->ringbuffer[0]->size);
-		memset(jd->ringbuffer[1]->buf, 0, jd->ringbuffer[1]->size);
-
-		port_name = xmalloc(sizeof(char)*(7+strlen(jd->name)));
-
-		sprintf(port_name, "%s:left", jd->name);
-		if ( (jack_connect(jd->client, port_name,
-				   jd->output_ports[0])) != 0 ) {
-			ERROR("%s is not a valid Jack Client / Port\n",
-			      jd->output_ports[0]);
-			free(port_name);
-			return -1;
-		}
-		sprintf(port_name, "%s:right", jd->name);
-		if ( (jack_connect(jd->client, port_name,
-				   jd->output_ports[1])) != 0 ) {
-			ERROR("%s is not a valid Jack Client / Port\n",
-			      jd->output_ports[1]);
-			free(port_name);
-			return -1;
-		}
-		free(port_name);
-	}
-
-	return 1;
-}
-
-static int
-mpd_jack_open(void *data, struct audio_format *audio_format)
-{
-	struct jack_data *jd = data;
-
-	assert(jd != NULL);
-
-	if (jd->client == NULL && mpd_jack_connect(jd, audio_format) < 0) {
-		mpd_jack_client_free(jd);
-		return -1;
-	}
-
-	set_audioformat(jd, audio_format);
-
-	return 0;
-}
-
-static void
-mpd_jack_close(mpd_unused void *data)
-{
-	/*mpd_jack_finish(audioOutput);*/
-}
-
-static void
-mpd_jack_cancel (mpd_unused void *data)
-{
-}
-
-static inline jack_default_audio_sample_t
-sample_16_to_jack(int16_t sample)
-{
-	return sample / (jack_default_audio_sample_t)(1 << (16 - 1));
-}
-
-static void
-mpd_jack_write_samples_16(struct jack_data *jd, const int16_t *src,
-			  unsigned num_samples)
-{
-	jack_default_audio_sample_t sample;
-
-	while (num_samples-- > 0) {
-		sample = sample_16_to_jack(*src++);
-		jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
-				      sizeof(sample));
-
-		sample = sample_16_to_jack(*src++);
-		jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
-				      sizeof(sample));
-	}
-}
-
-static inline jack_default_audio_sample_t
-sample_24_to_jack(int32_t sample)
-{
-	return sample / (jack_default_audio_sample_t)(1 << (24 - 1));
-}
-
-static void
-mpd_jack_write_samples_24(struct jack_data *jd, const int32_t *src,
-			  unsigned num_samples)
-{
-	jack_default_audio_sample_t sample;
-
-	while (num_samples-- > 0) {
-		sample = sample_24_to_jack(*src++);
-		jack_ringbuffer_write(jd->ringbuffer[0], (void*)&sample,
-				      sizeof(sample));
-
-		sample = sample_24_to_jack(*src++);
-		jack_ringbuffer_write(jd->ringbuffer[1], (void*)&sample,
-				      sizeof(sample));
-	}
-}
-
-static void
-mpd_jack_write_samples(struct jack_data *jd, const void *src,
-		       unsigned num_samples)
-{
-	switch (jd->audio_format->bits) {
-	case 16:
-		mpd_jack_write_samples_16(jd, (const int16_t*)src,
-					  num_samples);
-		break;
-
-	case 24:
-		mpd_jack_write_samples_24(jd, (const int32_t*)src,
-					  num_samples);
-		break;
-
-	default:
-		assert(false);
-	}
-}
-
-static int
-mpd_jack_play(void *data, const char *buff, size_t size)
-{
-	struct jack_data *jd = data;
-	const size_t frame_size = audio_format_frame_size(jd->audio_format);
-	size_t space, space1;
-
-	if (jd->shutdown) {
-		ERROR("Refusing to play, because there is no client thread.\n");
-		mpd_jack_client_free(jd);
-		audio_output_closed(jd->ao);
-		return 0;
-	}
-
-	assert(size % frame_size == 0);
-	size /= frame_size;
-	while (size > 0 && !jd->shutdown) {
-		space = jack_ringbuffer_write_space(jd->ringbuffer[0]);
-		space1 = jack_ringbuffer_write_space(jd->ringbuffer[1]);
-		if (space > space1)
-			/* send data symmetrically */
-			space = space1;
-
-		space /= sample_size;
-		if (space > 0) {
-			if (space > size)
-				space = size;
-
-			mpd_jack_write_samples(jd, buff, space);
-
-			buff += space * frame_size;
-			size -= space;
-		} else {
-			/* XXX do something more intelligent to
-			   synchronize */
-			my_usleep(10000);
-		}
-
-	}
-
-	return 0;
-}
-
-const struct audio_output_plugin jackPlugin = {
-	.name = "jack",
-	.test_default_device = mpd_jack_test_default_device,
-	.init = mpd_jack_init,
-	.finish = mpd_jack_finish,
-	.open = mpd_jack_open,
-	.play = mpd_jack_play,
-	.cancel = mpd_jack_cancel,
-	.close = mpd_jack_close,
-};
-
-#else /* HAVE JACK */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(jackPlugin)
-
-#endif /* HAVE_JACK */
diff --git a/src/audioOutputs/audioOutput_mvp.c b/src/audioOutputs/audioOutput_mvp.c
deleted file mode 100644
index 70dd25f9d..000000000
--- a/src/audioOutputs/audioOutput_mvp.c
+++ /dev/null
@@ -1,280 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * Media MVP audio output based on code from MVPMC project:
- * http://mvpmc.sourceforge.net/
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_MVP
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <sys/types.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-
-typedef struct {
-	unsigned long dsp_status;
-	unsigned long stream_decode_type;
-	unsigned long sample_rate;
-	unsigned long bit_rate;
-	unsigned long raw[64 / sizeof(unsigned long)];
-} aud_status_t;
-
-#define MVP_SET_AUD_STOP		_IOW('a',1,int)
-#define MVP_SET_AUD_PLAY		_IOW('a',2,int)
-#define MVP_SET_AUD_PAUSE	_IOW('a',3,int)
-#define MVP_SET_AUD_UNPAUSE	_IOW('a',4,int)
-#define MVP_SET_AUD_SRC		_IOW('a',5,int)
-#define MVP_SET_AUD_MUTE		_IOW('a',6,int)
-#define MVP_SET_AUD_BYPASS	_IOW('a',8,int)
-#define MVP_SET_AUD_CHANNEL	_IOW('a',9,int)
-#define MVP_GET_AUD_STATUS	_IOR('a',10,aud_status_t)
-#define MVP_SET_AUD_VOLUME	_IOW('a',13,int)
-#define MVP_GET_AUD_VOLUME	_IOR('a',14,int)
-#define MVP_SET_AUD_STREAMTYPE	_IOW('a',15,int)
-#define MVP_SET_AUD_FORMAT	_IOW('a',16,int)
-#define MVP_GET_AUD_SYNC		_IOR('a',21,pts_sync_data_t*)
-#define MVP_SET_AUD_STC		_IOW('a',22,long long int *)
-#define MVP_SET_AUD_SYNC		_IOW('a',23,int)
-#define MVP_SET_AUD_END_STREAM	_IOW('a',25,int)
-#define MVP_SET_AUD_RESET	_IOW('a',26,int)
-#define MVP_SET_AUD_DAC_CLK	_IOW('a',27,int)
-#define MVP_GET_AUD_REGS		_IOW('a',28,aud_ctl_regs_t*)
-
-typedef struct _MvpData {
-	struct audio_output *audio_output;
-	struct audio_format audio_format;
-	int fd;
-} MvpData;
-
-static unsigned pcmfrequencies[][3] = {
-	{9, 8000, 32000},
-	{10, 11025, 44100},
-	{11, 12000, 48000},
-	{1, 16000, 32000},
-	{2, 22050, 44100},
-	{3, 24000, 48000},
-	{5, 32000, 32000},
-	{0, 44100, 44100},
-	{7, 48000, 48000},
-	{13, 64000, 32000},
-	{14, 88200, 44100},
-	{15, 96000, 48000}
-};
-
-static const unsigned numfrequencies =
-	sizeof(pcmfrequencies) / sizeof(pcmfrequencies[0]);
-
-static int mvp_testDefault(void)
-{
-	int fd;
-
-	fd = open("/dev/adec_pcm", O_WRONLY);
-
-	if (fd) {
-		close(fd);
-		return 0;
-	}
-
-	WARNING("Error opening PCM device \"/dev/adec_pcm\": %s\n",
-		strerror(errno));
-
-	return -1;
-}
-
-static void *mvp_initDriver(mpd_unused struct audio_output *audio_output,
-			    mpd_unused const struct audio_format *audio_format,
-			    mpd_unused ConfigParam *param)
-{
-	MvpData *md = xmalloc(sizeof(MvpData));
-	md->audio_output = audio_output;
-	md->fd = -1;
-
-	return md;
-}
-
-static void mvp_finishDriver(void *data)
-{
-	MvpData *md = data;
-	free(md);
-}
-
-static int mvp_setPcmParams(MvpData * md, unsigned long rate, int channels,
-			    int big_endian, unsigned bits)
-{
-	unsigned iloop;
-	unsigned mix[5];
-
-	if (channels == 1)
-		mix[0] = 1;
-	else if (channels == 2)
-		mix[0] = 0;
-	else
-		return -1;
-
-	/* 0,1=24bit(24) , 2,3=16bit */
-	if (bits == 16)
-		mix[1] = 2;
-	else if (bits == 24)
-		mix[1] = 0;
-	else
-		return -1;
-
-	mix[3] = 0;	/* stream type? */
-
-	if (big_endian == 1)
-		mix[4] = 1;
-	else if (big_endian == 0)
-		mix[4] = 0;
-	else
-		return -1;
-
-	/*
-	 * if there is an exact match for the frequency, use it.
-	 */
-	for (iloop = 0; iloop < numfrequencies; iloop++) {
-		if (rate == pcmfrequencies[iloop][1]) {
-			mix[2] = pcmfrequencies[iloop][0];
-			break;
-		}
-	}
-
-	if (iloop >= numfrequencies) {
-		ERROR("Can not find suitable output frequency for %ld\n", rate);
-		return -1;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
-		ERROR("Can not set audio format\n");
-		return -1;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_SYNC, 2) != 0) {
-		ERROR("Can not set audio sync\n");
-		return -1;
-	}
-
-	if (ioctl(md->fd, MVP_SET_AUD_PLAY, 0) < 0) {
-		ERROR("Can not set audio play mode\n");
-		return -1;
-	}
-
-	return 0;
-}
-
-static int mvp_openDevice(void *data, struct audio_format *audioFormat)
-{
-	MvpData *md = data;
-	long long int stc = 0;
-	int mix[5] = { 0, 2, 7, 1, 0 };
-
-	if ((md->fd = open("/dev/adec_pcm", O_RDWR | O_NONBLOCK)) < 0) {
-		ERROR("Error opening /dev/adec_pcm: %s\n", strerror(errno));
-		return -1;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_SRC, 1) < 0) {
-		ERROR("Error setting audio source: %s\n", strerror(errno));
-		return -1;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_STREAMTYPE, 0) < 0) {
-		ERROR("Error setting audio streamtype: %s\n", strerror(errno));
-		return -1;
-	}
-	if (ioctl(md->fd, MVP_SET_AUD_FORMAT, &mix) < 0) {
-		ERROR("Error setting audio format: %s\n", strerror(errno));
-		return -1;
-	}
-	ioctl(md->fd, MVP_SET_AUD_STC, &stc);
-	if (ioctl(md->fd, MVP_SET_AUD_BYPASS, 1) < 0) {
-		ERROR("Error setting audio streamtype: %s\n", strerror(errno));
-		return -1;
-	}
-#ifdef WORDS_BIGENDIAN
-	mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
-			 0, audioFormat->bits);
-#else
-	mvp_setPcmParams(md, audioFormat->sample_rate, audioFormat->channels,
-			 1, audioFormat->bits);
-#endif
-	md->audio_format = *audioFormat;
-	return 0;
-}
-
-static void mvp_closeDevice(void *data)
-{
-	MvpData *md = data;
-	if (md->fd >= 0)
-		close(md->fd);
-	md->fd = -1;
-}
-
-static void mvp_dropBufferedAudio(void *data)
-{
-	MvpData *md = data;
-	if (md->fd >= 0) {
-		ioctl(md->fd, MVP_SET_AUD_RESET, 0x11);
-		close(md->fd);
-		md->fd = -1;
-		audio_output_closed(md->audio_output);
-	}
-}
-
-static int mvp_playAudio(void *data, const char *playChunk, size_t size)
-{
-	MvpData *md = data;
-	ssize_t ret;
-
-	/* reopen the device since it was closed by dropBufferedAudio */
-	if (md->fd < 0)
-		mvp_openDevice(md, &md->audio_format);
-
-	while (size > 0) {
-		ret = write(md->fd, playChunk, size);
-		if (ret < 0) {
-			if (errno == EINTR)
-				continue;
-			ERROR("closing mvp PCM device due to write error: "
-			      "%s\n", strerror(errno));
-			mvp_closeDevice(md);
-			return -1;
-		}
-		playChunk += ret;
-		size -= ret;
-	}
-	return 0;
-}
-
-const struct audio_output_plugin mvpPlugin = {
-	.name = "mvp",
-	.test_default_device = mvp_testDefault,
-	.init = mvp_initDriver,
-	.finish = mvp_finishDriver,
-	.open = mvp_openDevice,
-	.play = mvp_playAudio,
-	.cancel = mvp_dropBufferedAudio,
-	.close = mvp_closeDevice,
-};
-
-#else /* HAVE_MVP */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(mvpPlugin)
-#endif /* HAVE_MVP */
diff --git a/src/audioOutputs/audioOutput_null.c b/src/audioOutputs/audioOutput_null.c
deleted file mode 100644
index ff3a9833c..000000000
--- a/src/audioOutputs/audioOutput_null.c
+++ /dev/null
@@ -1,85 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-#include "../timer.h"
-#include "../utils.h"
-
-struct null_data {
-	Timer *timer;
-};
-
-static void *null_initDriver(mpd_unused struct audio_output *audioOutput,
-			     mpd_unused const struct audio_format *audio_format,
-			     mpd_unused ConfigParam *param)
-{
-	struct null_data *nd = xmalloc(sizeof(*nd));
-	nd->timer = NULL;
-	return nd;
-}
-
-static int null_openDevice(void *data,
-			   struct audio_format *audio_format)
-{
-	struct null_data *nd = data;
-
-	nd->timer = timer_new(audio_format);
-	return 0;
-}
-
-static void null_closeDevice(void *data)
-{
-	struct null_data *nd = data;
-
-	if (nd->timer != NULL) {
-		timer_free(nd->timer);
-		nd->timer = NULL;
-	}
-}
-
-static int null_playAudio(void *data,
-			  mpd_unused const char *playChunk, size_t size)
-{
-	struct null_data *nd = data;
-	Timer *timer = nd->timer;
-
-	if (!timer->started)
-		timer_start(timer);
-	else
-		timer_sync(timer);
-
-	timer_add(timer, size);
-
-	return 0;
-}
-
-static void null_dropBufferedAudio(void *data)
-{
-	struct null_data *nd = data;
-
-	timer_reset(nd->timer);
-}
-
-const struct audio_output_plugin nullPlugin = {
-	.name = "null",
-	.init = null_initDriver,
-	.open = null_openDevice,
-	.play = null_playAudio,
-	.cancel = null_dropBufferedAudio,
-	.close = null_closeDevice,
-};
diff --git a/src/audioOutputs/audioOutput_oss.c b/src/audioOutputs/audioOutput_oss.c
deleted file mode 100644
index 67ea11fe4..000000000
--- a/src/audioOutputs/audioOutput_oss.c
+++ /dev/null
@@ -1,571 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * OSS audio output (c) 2004, 2005, 2006, 2007 by Eric Wong <eric@petta-tech.com>
- *                   and Warren Dukes <warren.dukes@gmail.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_OSS
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <fcntl.h>
-
-#if defined(__OpenBSD__) || defined(__NetBSD__)
-# include <soundcard.h>
-#else /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
-# include <sys/soundcard.h>
-#endif /* !(defined(__OpenBSD__) || defined(__NetBSD__) */
-
-#ifdef WORDS_BIGENDIAN
-# define	AFMT_S16_MPD	 AFMT_S16_BE
-#else
-# define	AFMT_S16_MPD	 AFMT_S16_LE
-#endif /* WORDS_BIGENDIAN */
-
-typedef struct _OssData {
-	int fd;
-	const char *device;
-	struct audio_format audio_format;
-	int bitFormat;
-	int *supported[3];
-	int numSupported[3];
-	int *unsupported[3];
-	int numUnsupported[3];
-} OssData;
-
-enum oss_support {
-	OSS_SUPPORTED = 1,
-	OSS_UNSUPPORTED = 0,
-	OSS_UNKNOWN = -1,
-};
-
-enum oss_param {
-	OSS_RATE = 0,
-	OSS_CHANNELS = 1,
-	OSS_BITS = 2,
-};
-
-static enum oss_param
-getIndexForParam(unsigned param)
-{
-	enum oss_param idx = OSS_RATE;
-
-	switch (param) {
-	case SNDCTL_DSP_SPEED:
-		idx = OSS_RATE;
-		break;
-	case SNDCTL_DSP_CHANNELS:
-		idx = OSS_CHANNELS;
-		break;
-	case SNDCTL_DSP_SAMPLESIZE:
-		idx = OSS_BITS;
-		break;
-	}
-
-	return idx;
-}
-
-static int findSupportedParam(OssData * od, unsigned param, int val)
-{
-	int i;
-	enum oss_param idx = getIndexForParam(param);
-
-	for (i = 0; i < od->numSupported[idx]; i++) {
-		if (od->supported[idx][i] == val)
-			return 1;
-	}
-
-	return 0;
-}
-
-static int canConvert(int idx, int val)
-{
-	switch (idx) {
-	case OSS_BITS:
-		if (val != 16)
-			return 0;
-		break;
-	case OSS_CHANNELS:
-		if (val != 2)
-			return 0;
-		break;
-	}
-
-	return 1;
-}
-
-static int getSupportedParam(OssData * od, unsigned param, int val)
-{
-	int i;
-	enum oss_param idx = getIndexForParam(param);
-	int ret = -1;
-	int least = val;
-	int diff;
-
-	for (i = 0; i < od->numSupported[idx]; i++) {
-		diff = od->supported[idx][i] - val;
-		if (diff < 0)
-			diff = -diff;
-		if (diff < least) {
-			if (!canConvert(idx, od->supported[idx][i])) {
-				continue;
-			}
-			least = diff;
-			ret = od->supported[idx][i];
-		}
-	}
-
-	return ret;
-}
-
-static int findUnsupportedParam(OssData * od, unsigned param, int val)
-{
-	int i;
-	enum oss_param idx = getIndexForParam(param);
-
-	for (i = 0; i < od->numUnsupported[idx]; i++) {
-		if (od->unsupported[idx][i] == val)
-			return 1;
-	}
-
-	return 0;
-}
-
-static void addSupportedParam(OssData * od, unsigned param, int val)
-{
-	enum oss_param idx = getIndexForParam(param);
-
-	od->numSupported[idx]++;
-	od->supported[idx] = xrealloc(od->supported[idx],
-				      od->numSupported[idx] * sizeof(int));
-	od->supported[idx][od->numSupported[idx] - 1] = val;
-}
-
-static void addUnsupportedParam(OssData * od, unsigned param, int val)
-{
-	enum oss_param idx = getIndexForParam(param);
-
-	od->numUnsupported[idx]++;
-	od->unsupported[idx] = xrealloc(od->unsupported[idx],
-					od->numUnsupported[idx] *
-					sizeof(int));
-	od->unsupported[idx][od->numUnsupported[idx] - 1] = val;
-}
-
-static void removeSupportedParam(OssData * od, unsigned param, int val)
-{
-	int i;
-	int j = 0;
-	enum oss_param idx = getIndexForParam(param);
-
-	for (i = 0; i < od->numSupported[idx] - 1; i++) {
-		if (od->supported[idx][i] == val)
-			j = 1;
-		od->supported[idx][i] = od->supported[idx][i + j];
-	}
-
-	od->numSupported[idx]--;
-	od->supported[idx] = xrealloc(od->supported[idx],
-				      od->numSupported[idx] * sizeof(int));
-}
-
-static void removeUnsupportedParam(OssData * od, unsigned param, int val)
-{
-	int i;
-	int j = 0;
-	enum oss_param idx = getIndexForParam(param);
-
-	for (i = 0; i < od->numUnsupported[idx] - 1; i++) {
-		if (od->unsupported[idx][i] == val)
-			j = 1;
-		od->unsupported[idx][i] = od->unsupported[idx][i + j];
-	}
-
-	od->numUnsupported[idx]--;
-	od->unsupported[idx] = xrealloc(od->unsupported[idx],
-					od->numUnsupported[idx] *
-					sizeof(int));
-}
-
-static enum oss_support
-isSupportedParam(OssData * od, unsigned param, int val)
-{
-	if (findSupportedParam(od, param, val))
-		return OSS_SUPPORTED;
-	if (findUnsupportedParam(od, param, val))
-		return OSS_UNSUPPORTED;
-	return OSS_UNKNOWN;
-}
-
-static void supportParam(OssData * od, unsigned param, int val)
-{
-	enum oss_support supported = isSupportedParam(od, param, val);
-
-	if (supported == OSS_SUPPORTED)
-		return;
-
-	if (supported == OSS_UNSUPPORTED) {
-		removeUnsupportedParam(od, param, val);
-	}
-
-	addSupportedParam(od, param, val);
-}
-
-static void unsupportParam(OssData * od, unsigned param, int val)
-{
-	enum oss_support supported = isSupportedParam(od, param, val);
-
-	if (supported == OSS_UNSUPPORTED)
-		return;
-
-	if (supported == OSS_SUPPORTED) {
-		removeSupportedParam(od, param, val);
-	}
-
-	addUnsupportedParam(od, param, val);
-}
-
-static OssData *newOssData(void)
-{
-	OssData *ret = xmalloc(sizeof(OssData));
-
-	ret->device = NULL;
-	ret->fd = -1;
-
-	ret->supported[OSS_RATE] = NULL;
-	ret->supported[OSS_CHANNELS] = NULL;
-	ret->supported[OSS_BITS] = NULL;
-	ret->unsupported[OSS_RATE] = NULL;
-	ret->unsupported[OSS_CHANNELS] = NULL;
-	ret->unsupported[OSS_BITS] = NULL;
-
-	ret->numSupported[OSS_RATE] = 0;
-	ret->numSupported[OSS_CHANNELS] = 0;
-	ret->numSupported[OSS_BITS] = 0;
-	ret->numUnsupported[OSS_RATE] = 0;
-	ret->numUnsupported[OSS_CHANNELS] = 0;
-	ret->numUnsupported[OSS_BITS] = 0;
-
-	supportParam(ret, SNDCTL_DSP_SPEED, 48000);
-	supportParam(ret, SNDCTL_DSP_SPEED, 44100);
-	supportParam(ret, SNDCTL_DSP_CHANNELS, 2);
-	supportParam(ret, SNDCTL_DSP_SAMPLESIZE, 16);
-
-	return ret;
-}
-
-static void freeOssData(OssData * od)
-{
-	if (od->supported[OSS_RATE])
-		free(od->supported[OSS_RATE]);
-	if (od->supported[OSS_CHANNELS])
-		free(od->supported[OSS_CHANNELS]);
-	if (od->supported[OSS_BITS])
-		free(od->supported[OSS_BITS]);
-	if (od->unsupported[OSS_RATE])
-		free(od->unsupported[OSS_RATE]);
-	if (od->unsupported[OSS_CHANNELS])
-		free(od->unsupported[OSS_CHANNELS]);
-	if (od->unsupported[OSS_BITS])
-		free(od->unsupported[OSS_BITS]);
-
-	free(od);
-}
-
-#define OSS_STAT_NO_ERROR 	0
-#define OSS_STAT_NOT_CHAR_DEV	-1
-#define OSS_STAT_NO_PERMS	-2
-#define OSS_STAT_DOESN_T_EXIST	-3
-#define OSS_STAT_OTHER		-4
-
-static int oss_statDevice(const char *device, int *stErrno)
-{
-	struct stat st;
-
-	if (0 == stat(device, &st)) {
-		if (!S_ISCHR(st.st_mode)) {
-			return OSS_STAT_NOT_CHAR_DEV;
-		}
-	} else {
-		*stErrno = errno;
-
-		switch (errno) {
-		case ENOENT:
-		case ENOTDIR:
-			return OSS_STAT_DOESN_T_EXIST;
-		case EACCES:
-			return OSS_STAT_NO_PERMS;
-		default:
-			return OSS_STAT_OTHER;
-		}
-	}
-
-	return 0;
-}
-
-static const char *default_devices[] = { "/dev/sound/dsp", "/dev/dsp" };
-
-static int oss_testDefault(void)
-{
-	int fd, i;
-
-	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
-		if ((fd = open(default_devices[i], O_WRONLY)) >= 0) {
-			xclose(fd);
-			return 0;
-		}
-		WARNING("Error opening OSS device \"%s\": %s\n",
-		        default_devices[i], strerror(errno));
-	}
-
-	return -1;
-}
-
-static void *oss_open_default(ConfigParam *param)
-{
-	int i;
-	int err[ARRAY_SIZE(default_devices)];
-	int ret[ARRAY_SIZE(default_devices)];
-
-	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
-		ret[i] = oss_statDevice(default_devices[i], &err[i]);
-		if (ret[i] == 0) {
-			OssData *od = newOssData();
-			od->device = default_devices[i];
-			return od;
-		}
-	}
-
-	if (param)
-		ERROR("error trying to open specified OSS device"
-	              " at line %i\n", param->line);
-	else
-		ERROR("error trying to open default OSS device\n");
-
-	for (i = ARRAY_SIZE(default_devices); --i >= 0; ) {
-		const char *dev = default_devices[i];
-		switch(ret[i]) {
-		case OSS_STAT_DOESN_T_EXIST:
-			ERROR("%s not found\n", dev);
-			break;
-		case OSS_STAT_NOT_CHAR_DEV:
-			ERROR("%s is not a character device\n", dev);
-			break;
-		case OSS_STAT_NO_PERMS:
-			ERROR("%s: permission denied\n", dev);
-			break;
-		default:
-			ERROR("Error accessing %s: %s\n", dev, strerror(err[i]));
-		}
-	}
-	exit(EXIT_FAILURE);
-	return NULL; /* some compilers can be dumb... */
-}
-
-static void *oss_initDriver(mpd_unused struct audio_output *audioOutput,
-			    mpd_unused const struct audio_format *audio_format,
-			    ConfigParam * param)
-{
-	if (param) {
-		BlockParam *bp = getBlockParam(param, "device");
-		if (bp) {
-			OssData *od = newOssData();
-			od->device = bp->value;
-			return od;
-		}
-	}
-	return oss_open_default(param);
-}
-
-static void oss_finishDriver(void *data)
-{
-	OssData *od = data;
-
-	freeOssData(od);
-}
-
-static int setParam(OssData * od, unsigned param, int *value)
-{
-	int val = *value;
-	int copy;
-	enum oss_support supported = isSupportedParam(od, param, val);
-
-	do {
-		if (supported == OSS_UNSUPPORTED) {
-			val = getSupportedParam(od, param, val);
-			if (copy < 0)
-				return -1;
-		}
-		copy = val;
-		if (ioctl(od->fd, param, &copy)) {
-			unsupportParam(od, param, val);
-			supported = OSS_UNSUPPORTED;
-		} else {
-			if (supported == OSS_UNKNOWN) {
-				supportParam(od, param, val);
-				supported = OSS_SUPPORTED;
-			}
-			val = copy;
-		}
-	} while (supported == OSS_UNSUPPORTED);
-
-	*value = val;
-
-	return 0;
-}
-
-static void oss_close(OssData * od)
-{
-	if (od->fd >= 0)
-		while (close(od->fd) && errno == EINTR) ;
-	od->fd = -1;
-}
-
-static int oss_open(OssData *od)
-{
-	int tmp;
-
-	if ((od->fd = open(od->device, O_WRONLY)) < 0) {
-		ERROR("Error opening OSS device \"%s\": %s\n", od->device,
-		      strerror(errno));
-		goto fail;
-	}
-
-	tmp = od->audio_format.channels;
-	if (setParam(od, SNDCTL_DSP_CHANNELS, &tmp)) {
-		ERROR("OSS device \"%s\" does not support %u channels: %s\n",
-		      od->device, od->audio_format.channels, strerror(errno));
-		goto fail;
-	}
-	od->audio_format.channels = tmp;
-
-	tmp = od->audio_format.sample_rate;
-	if (setParam(od, SNDCTL_DSP_SPEED, &tmp)) {
-		ERROR("OSS device \"%s\" does not support %u Hz audio: %s\n",
-		      od->device, od->audio_format.sample_rate,
-		      strerror(errno));
-		goto fail;
-	}
-	od->audio_format.sample_rate = tmp;
-
-	switch (od->audio_format.bits) {
-	case 8:
-		tmp = AFMT_S8;
-		break;
-	case 16:
-		tmp = AFMT_S16_MPD;
-	}
-
-	if (setParam(od, SNDCTL_DSP_SAMPLESIZE, &tmp)) {
-		ERROR("OSS device \"%s\" does not support %u bit audio: %s\n",
-		      od->device, tmp, strerror(errno));
-		goto fail;
-	}
-
-	return 0;
-
-fail:
-	oss_close(od);
-	return -1;
-}
-
-static int oss_openDevice(void *data,
-			  struct audio_format *audioFormat)
-{
-	int ret;
-	OssData *od = data;
-
-	od->audio_format = *audioFormat;
-
-	if ((ret = oss_open(od)) < 0)
-		return ret;
-
-	*audioFormat = od->audio_format;
-
-	DEBUG("oss device \"%s\" will be playing %u bit %u channel audio at "
-	      "%u Hz\n", od->device,
-	      od->audio_format.bits, od->audio_format.channels,
-	      od->audio_format.sample_rate);
-
-	return ret;
-}
-
-static void oss_closeDevice(void *data)
-{
-	OssData *od = data;
-
-	oss_close(od);
-}
-
-static void oss_dropBufferedAudio(void *data)
-{
-	OssData *od = data;
-
-	if (od->fd >= 0) {
-		ioctl(od->fd, SNDCTL_DSP_RESET, 0);
-		oss_close(od);
-	}
-}
-
-static int oss_playAudio(void *data,
-			 const char *playChunk, size_t size)
-{
-	OssData *od = data;
-	ssize_t ret;
-
-	/* reopen the device since it was closed by dropBufferedAudio */
-	if (od->fd < 0 && oss_open(od) < 0)
-		return -1;
-
-	while (size > 0) {
-		ret = write(od->fd, playChunk, size);
-		if (ret < 0) {
-			if (errno == EINTR)
-				continue;
-			ERROR("closing oss device \"%s\" due to write error: "
-			      "%s\n", od->device, strerror(errno));
-			oss_closeDevice(od);
-			return -1;
-		}
-		playChunk += ret;
-		size -= ret;
-	}
-
-	return 0;
-}
-
-const struct audio_output_plugin ossPlugin = {
-	.name = "oss",
-	.test_default_device = oss_testDefault,
-	.init = oss_initDriver,
-	.finish = oss_finishDriver,
-	.open = oss_openDevice,
-	.play = oss_playAudio,
-	.cancel = oss_dropBufferedAudio,
-	.close = oss_closeDevice,
-};
-
-#else /* HAVE OSS */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(ossPlugin)
-#endif /* HAVE_OSS */
diff --git a/src/audioOutputs/audioOutput_osx.c b/src/audioOutputs/audioOutput_osx.c
deleted file mode 100644
index 65060cc8c..000000000
--- a/src/audioOutputs/audioOutput_osx.c
+++ /dev/null
@@ -1,368 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_OSX
-
-#include <AudioUnit/AudioUnit.h>
-
-#include "../utils.h"
-#include "../log.h"
-
-typedef struct _OsxData {
-	AudioUnit au;
-	pthread_mutex_t mutex;
-	pthread_cond_t condition;
-	char *buffer;
-	size_t bufferSize;
-	size_t pos;
-	size_t len;
-	int started;
-} OsxData;
-
-static OsxData *newOsxData()
-{
-	OsxData *ret = xmalloc(sizeof(OsxData));
-
-	pthread_mutex_init(&ret->mutex, NULL);
-	pthread_cond_init(&ret->condition, NULL);
-
-	ret->pos = 0;
-	ret->len = 0;
-	ret->started = 0;
-	ret->buffer = NULL;
-	ret->bufferSize = 0;
-
-	return ret;
-}
-
-static int osx_testDefault()
-{
-	/*AudioUnit au;
-	   ComponentDescription desc;
-	   Component comp;
-
-	   desc.componentType = kAudioUnitType_Output;
-	   desc.componentSubType = kAudioUnitSubType_Output;
-	   desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-	   desc.componentFlags = 0;
-	   desc.componentFlagsMask = 0;
-
-	   comp = FindNextComponent(NULL, &desc);
-	   if(!comp) {
-	   ERROR("Unable to open default OS X defice\n");
-	   return -1;
-	   }
-
-	   if(OpenAComponent(comp, &au) != noErr) {
-	   ERROR("Unable to open default OS X defice\n");
-	   return -1;
-	   }
-
-	   CloseComponent(au); */
-
-	return 0;
-}
-
-static int osx_initDriver(struct audio_output *audioOutput,
-			  mpd_unused const struct audio_format *audio_format,
-			  ConfigParam * param)
-{
-	OsxData *od = newOsxData();
-
-	audioOutput->data = od;
-
-	return 0;
-}
-
-static void freeOsxData(OsxData * od)
-{
-	if (od->buffer)
-		free(od->buffer);
-	pthread_mutex_destroy(&od->mutex);
-	pthread_cond_destroy(&od->condition);
-	free(od);
-}
-
-static void osx_finishDriver(struct audio_output *audioOutput)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-	freeOsxData(od);
-}
-
-static void osx_dropBufferedAudio(struct audio_output *audioOutput)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-
-	pthread_mutex_lock(&od->mutex);
-	od->len = 0;
-	pthread_mutex_unlock(&od->mutex);
-}
-
-static void osx_closeDevice(struct audio_output *audioOutput)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-
-	pthread_mutex_lock(&od->mutex);
-	while (od->len) {
-		pthread_cond_wait(&od->condition, &od->mutex);
-	}
-	pthread_mutex_unlock(&od->mutex);
-
-	if (od->started) {
-		AudioOutputUnitStop(od->au);
-		od->started = 0;
-	}
-
-	CloseComponent(od->au);
-	AudioUnitUninitialize(od->au);
-}
-
-static OSStatus osx_render(void *vdata,
-			   AudioUnitRenderActionFlags * ioActionFlags,
-			   const AudioTimeStamp * inTimeStamp,
-			   UInt32 inBusNumber, UInt32 inNumberFrames,
-			   AudioBufferList * bufferList)
-{
-	OsxData *od = (OsxData *) vdata;
-	AudioBuffer *buffer = &bufferList->mBuffers[0];
-	size_t bufferSize = buffer->mDataByteSize;
-	size_t bytesToCopy;
-	int curpos = 0;
-
-	/*DEBUG("osx_render: enter : %i\n", (int)bufferList->mNumberBuffers);
-	   DEBUG("osx_render: ioActionFlags: %p\n", ioActionFlags);
-	   if(ioActionFlags) {
-	   if(*ioActionFlags & kAudioUnitRenderAction_PreRender) {
-	   DEBUG("prerender\n");
-	   }
-	   if(*ioActionFlags & kAudioUnitRenderAction_PostRender) {
-	   DEBUG("post render\n");
-	   }
-	   if(*ioActionFlags & kAudioUnitRenderAction_OutputIsSilence) {
-	   DEBUG("post render\n");
-	   }
-	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Preflight) {
-	   DEBUG("prefilight\n");
-	   }
-	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Render) {
-	   DEBUG("render\n");
-	   }
-	   if(*ioActionFlags & kAudioOfflineUnitRenderAction_Complete) {
-	   DEBUG("complete\n");
-	   }
-	   } */
-
-	/* while(bufferSize) {
-	   DEBUG("osx_render: lock\n"); */
-	pthread_mutex_lock(&od->mutex);
-	/*
-	   DEBUG("%i:%i\n", bufferSize, od->len);
-	   while(od->go && od->len < bufferSize && 
-	   od->len < od->bufferSize)
-	   {
-	   DEBUG("osx_render: wait\n");
-	   pthread_cond_wait(&od->condition, &od->mutex);
-	   }
-	 */
-
-	bytesToCopy = od->len < bufferSize ? od->len : bufferSize;
-	bufferSize = bytesToCopy;
-	od->len -= bytesToCopy;
-
-	if (od->pos + bytesToCopy > od->bufferSize) {
-		size_t bytes = od->bufferSize - od->pos;
-		memcpy(buffer->mData + curpos, od->buffer + od->pos, bytes);
-		od->pos = 0;
-		curpos += bytes;
-		bytesToCopy -= bytes;
-	}
-
-	memcpy(buffer->mData + curpos, od->buffer + od->pos, bytesToCopy);
-	od->pos += bytesToCopy;
-	curpos += bytesToCopy;
-
-	if (od->pos >= od->bufferSize)
-		od->pos = 0;
-	/* DEBUG("osx_render: unlock\n"); */
-	pthread_mutex_unlock(&od->mutex);
-	pthread_cond_signal(&od->condition);
-	/* } */
-
-	buffer->mDataByteSize = bufferSize;
-
-	if (!bufferSize) {
-		my_usleep(1000);
-	}
-
-	/* DEBUG("osx_render: leave\n"); */
-	return 0;
-}
-
-static int osx_openDevice(struct audio_output *audioOutput,
-			  struct audio_format *audioFormat)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-	ComponentDescription desc;
-	Component comp;
-	AURenderCallbackStruct callback;
-	AudioStreamBasicDescription streamDesc;
-
-	desc.componentType = kAudioUnitType_Output;
-	desc.componentSubType = kAudioUnitSubType_DefaultOutput;
-	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-	desc.componentFlags = 0;
-	desc.componentFlagsMask = 0;
-
-	comp = FindNextComponent(NULL, &desc);
-	if (comp == 0) {
-		ERROR("Error finding OS X component\n");
-		return -1;
-	}
-
-	if (OpenAComponent(comp, &od->au) != noErr) {
-		ERROR("Unable to open OS X component\n");
-		return -1;
-	}
-
-	if (AudioUnitInitialize(od->au) != 0) {
-		CloseComponent(od->au);
-		ERROR("Unable to initialize OS X audio unit\n");
-		return -1;
-	}
-
-	callback.inputProc = osx_render;
-	callback.inputProcRefCon = od;
-
-	if (AudioUnitSetProperty(od->au, kAudioUnitProperty_SetRenderCallback,
-				 kAudioUnitScope_Input, 0,
-				 &callback, sizeof(callback)) != 0) {
-		AudioUnitUninitialize(od->au);
-		CloseComponent(od->au);
-		ERROR("unable to set callback for OS X audio unit\n");
-		return -1;
-	}
-
-	streamDesc.mSampleRate = audioFormat->sample_rate;
-	streamDesc.mFormatID = kAudioFormatLinearPCM;
-	streamDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger;
-#ifdef WORDS_BIGENDIAN
-	streamDesc.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-#endif
-
-	streamDesc.mBytesPerPacket = audio_format_frame_size(audioFormat);
-	streamDesc.mFramesPerPacket = 1;
-	streamDesc.mBytesPerFrame = streamDesc.mBytesPerPacket;
-	streamDesc.mChannelsPerFrame = audioFormat->channels;
-	streamDesc.mBitsPerChannel = audioFormat->bits;
-
-	if (AudioUnitSetProperty(od->au, kAudioUnitProperty_StreamFormat,
-				 kAudioUnitScope_Input, 0,
-				 &streamDesc, sizeof(streamDesc)) != 0) {
-		AudioUnitUninitialize(od->au);
-		CloseComponent(od->au);
-		ERROR("Unable to set format on OS X device\n");
-		return -1;
-	}
-
-	/* create a buffer of 1s */
-	od->bufferSize = (audioFormat->sample_rate) *
-		audio_format_frame_size(audioFormat);
-	od->buffer = xrealloc(od->buffer, od->bufferSize);
-
-	od->pos = 0;
-	od->len = 0;
-
-	return 0;
-}
-
-static int osx_play(struct audio_output *audioOutput,
-		    const char *playChunk, size_t size)
-{
-	OsxData *od = (OsxData *) audioOutput->data;
-	size_t bytesToCopy;
-	size_t curpos;
-
-	/* DEBUG("osx_play: enter\n"); */
-
-	if (!od->started) {
-		int err;
-		od->started = 1;
-		err = AudioOutputUnitStart(od->au);
-		if (err) {
-			ERROR("unable to start audio output: %i\n", err);
-			return -1;
-		}
-	}
-
-	pthread_mutex_lock(&od->mutex);
-
-	while (size) {
-		/* DEBUG("osx_play: lock\n"); */
-		curpos = od->pos + od->len;
-		if (curpos >= od->bufferSize)
-			curpos -= od->bufferSize;
-
-		bytesToCopy = od->bufferSize < size ? od->bufferSize : size;
-
-		while (od->len > od->bufferSize - bytesToCopy) {
-			/* DEBUG("osx_play: wait\n"); */
-			pthread_cond_wait(&od->condition, &od->mutex);
-		}
-
-		bytesToCopy = od->bufferSize - od->len;
-		bytesToCopy = bytesToCopy < size ? bytesToCopy : size;
-		size -= bytesToCopy;
-		od->len += bytesToCopy;
-
-		if (curpos + bytesToCopy > od->bufferSize) {
-			size_t bytes = od->bufferSize - curpos;
-			memcpy(od->buffer + curpos, playChunk, bytes);
-			curpos = 0;
-			playChunk += bytes;
-			bytesToCopy -= bytes;
-		}
-
-		memcpy(od->buffer + curpos, playChunk, bytesToCopy);
-		curpos += bytesToCopy;
-		playChunk += bytesToCopy;
-
-	}
-	/* DEBUG("osx_play: unlock\n"); */
-	pthread_mutex_unlock(&od->mutex);
-
-	/* DEBUG("osx_play: leave\n"); */
-	return 0;
-}
-
-const struct audio_output_plugin osxPlugin = {
-	.name = "osx",
-	.test_default_device = osx_testDefault,
-	.init = osx_initDriver,
-	.finish = osx_finishDriver,
-	.open = osx_openDevice,
-	.play = osx_play,
-	.cancel = osx_dropBufferedAudio,
-	.close = osx_closeDevice,
-};
-
-#else
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(osxPlugin)
-#endif
diff --git a/src/audioOutputs/audioOutput_pulse.c b/src/audioOutputs/audioOutput_pulse.c
deleted file mode 100644
index b1ce3d049..000000000
--- a/src/audioOutputs/audioOutput_pulse.c
+++ /dev/null
@@ -1,218 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_PULSE
-
-#include "../utils.h"
-#include "../log.h"
-
-#include <pulse/simple.h>
-#include <pulse/error.h>
-
-#define MPD_PULSE_NAME "mpd"
-#define CONN_ATTEMPT_INTERVAL 60
-
-typedef struct _PulseData {
-	struct audio_output *ao;
-
-	pa_simple *s;
-	char *server;
-	char *sink;
-	int connAttempts;
-	time_t lastAttempt;
-} PulseData;
-
-static PulseData *newPulseData(void)
-{
-	PulseData *ret;
-
-	ret = xmalloc(sizeof(PulseData));
-
-	ret->s = NULL;
-	ret->server = NULL;
-	ret->sink = NULL;
-	ret->connAttempts = 0;
-	ret->lastAttempt = 0;
-
-	return ret;
-}
-
-static void freePulseData(PulseData * pd)
-{
-	if (pd->server)
-		free(pd->server);
-	if (pd->sink)
-		free(pd->sink);
-	free(pd);
-}
-
-static void *pulse_initDriver(struct audio_output *ao,
-			      mpd_unused const struct audio_format *audio_format,
-			      ConfigParam * param)
-{
-	BlockParam *server = NULL;
-	BlockParam *sink = NULL;
-	PulseData *pd;
-
-	if (param) {
-		server = getBlockParam(param, "server");
-		sink = getBlockParam(param, "sink");
-	}
-
-	pd = newPulseData();
-	pd->ao = ao;
-	pd->server = server ? xstrdup(server->value) : NULL;
-	pd->sink = sink ? xstrdup(sink->value) : NULL;
-
-	return pd;
-}
-
-static void pulse_finishDriver(void *data)
-{
-	PulseData *pd = data;
-
-	freePulseData(pd);
-}
-
-static int pulse_testDefault(void)
-{
-	pa_simple *s;
-	pa_sample_spec ss;
-	int error;
-
-	ss.format = PA_SAMPLE_S16NE;
-	ss.rate = 44100;
-	ss.channels = 2;
-
-	s = pa_simple_new(NULL, MPD_PULSE_NAME, PA_STREAM_PLAYBACK, NULL,
-			  MPD_PULSE_NAME, &ss, NULL, NULL, &error);
-	if (!s) {
-		WARNING("Cannot connect to default PulseAudio server: %s\n",
-			pa_strerror(error));
-		return -1;
-	}
-
-	pa_simple_free(s);
-
-	return 0;
-}
-
-static int pulse_openDevice(void *data,
-			    struct audio_format *audioFormat)
-{
-	PulseData *pd = data;
-	pa_sample_spec ss;
-	time_t t;
-	int error;
-
-	t = time(NULL);
-
-	if (pd->connAttempts != 0 &&
-	    (t - pd->lastAttempt) < CONN_ATTEMPT_INTERVAL)
-		return -1;
-
-	pd->connAttempts++;
-	pd->lastAttempt = t;
-
-	/* MPD doesn't support the other pulseaudio sample formats, so
-	   we just force MPD to send us everything as 16 bit */
-	audioFormat->bits = 16;
-
-	ss.format = PA_SAMPLE_S16NE;
-	ss.rate = audioFormat->sample_rate;
-	ss.channels = audioFormat->channels;
-
-	pd->s = pa_simple_new(pd->server, MPD_PULSE_NAME, PA_STREAM_PLAYBACK,
-			      pd->sink, audio_output_get_name(pd->ao),
-			      &ss, NULL, NULL,
-			      &error);
-	if (!pd->s) {
-		ERROR("Cannot connect to server in PulseAudio output "
-		      "\"%s\" (attempt %i): %s\n",
-		      audio_output_get_name(pd->ao),
-		      pd->connAttempts, pa_strerror(error));
-		return -1;
-	}
-
-	pd->connAttempts = 0;
-
-	DEBUG("PulseAudio output \"%s\" connected and playing %i bit, %i "
-	      "channel audio at %i Hz\n",
-	      audio_output_get_name(pd->ao),
-	      audioFormat->bits,
-	      audioFormat->channels, audioFormat->sample_rate);
-
-	return 0;
-}
-
-static void pulse_dropBufferedAudio(void *data)
-{
-	PulseData *pd = data;
-	int error;
-
-	if (pa_simple_flush(pd->s, &error) < 0)
-		WARNING("Flush failed in PulseAudio output \"%s\": %s\n",
-			audio_output_get_name(pd->ao),
-			pa_strerror(error));
-}
-
-static void pulse_closeDevice(void *data)
-{
-	PulseData *pd = data;
-
-	if (pd->s) {
-		pa_simple_drain(pd->s, NULL);
-		pa_simple_free(pd->s);
-	}
-}
-
-static int pulse_playAudio(void *data,
-			   const char *playChunk, size_t size)
-{
-	PulseData *pd = data;
-	int error;
-
-	if (pa_simple_write(pd->s, playChunk, size, &error) < 0) {
-		ERROR("PulseAudio output \"%s\" disconnecting due to write "
-		      "error: %s\n",
-		      audio_output_get_name(pd->ao),
-		      pa_strerror(error));
-		pulse_closeDevice(pd);
-		return -1;
-	}
-
-	return 0;
-}
-
-const struct audio_output_plugin pulsePlugin = {
-	.name = "pulse",
-	.test_default_device = pulse_testDefault,
-	.init = pulse_initDriver,
-	.finish = pulse_finishDriver,
-	.open = pulse_openDevice,
-	.play = pulse_playAudio,
-	.cancel = pulse_dropBufferedAudio,
-	.close = pulse_closeDevice,
-};
-
-#else /* HAVE_PULSE */
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(pulsePlugin)
-#endif /* HAVE_PULSE */
diff --git a/src/audioOutputs/audioOutput_shout.c b/src/audioOutputs/audioOutput_shout.c
deleted file mode 100644
index ae31b07ea..000000000
--- a/src/audioOutputs/audioOutput_shout.c
+++ /dev/null
@@ -1,596 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "audioOutput_shout.h"
-
-#ifdef HAVE_SHOUT
-
-#include "../utils.h"
-
-#include <assert.h>
-
-#define CONN_ATTEMPT_INTERVAL 60
-#define DEFAULT_CONN_TIMEOUT  2
-
-static int shout_init_count;
-
-static const struct shout_encoder_plugin *const shout_encoder_plugins[] = {
-#ifdef HAVE_SHOUT_MP3
-	&shout_mp3_encoder,
-#endif
-#ifdef HAVE_SHOUT_OGG
-	&shout_ogg_encoder,
-#endif
-	NULL
-};
-
-static const struct shout_encoder_plugin *
-shout_encoder_plugin_get(const char *name)
-{
-	unsigned i;
-
-	for (i = 0; shout_encoder_plugins[i] != NULL; ++i)
-		if (strcmp(shout_encoder_plugins[i]->name, name) == 0)
-			return shout_encoder_plugins[i];
-
-	return NULL;
-}
-
-static struct shout_data *new_shout_data(void)
-{
-	struct shout_data *ret = xmalloc(sizeof(*ret));
-
-	ret->shout_conn = shout_new();
-	ret->shout_meta = shout_metadata_new();
-	ret->opened = 0;
-	ret->tag = NULL;
-	ret->tag_to_send = 0;
-	ret->bitrate = -1;
-	ret->quality = -2.0;
-	ret->timeout = DEFAULT_CONN_TIMEOUT;
-	ret->conn_attempts = 0;
-	ret->last_attempt = 0;
-	ret->timer = NULL;
-	ret->buf.len = 0;
-
-	return ret;
-}
-
-static void free_shout_data(struct shout_data *sd)
-{
-	if (sd->shout_meta)
-		shout_metadata_free(sd->shout_meta);
-	if (sd->shout_conn)
-		shout_free(sd->shout_conn);
-	if (sd->tag)
-		tag_free(sd->tag);
-	if (sd->timer)
-		timer_free(sd->timer);
-
-	free(sd);
-}
-
-#define check_block_param(name) {		  \
-		block_param = getBlockParam(param, name);	\
-		if (!block_param) {					\
-			FATAL("no \"%s\" defined for shout device defined at line " \
-			      "%i\n", name, param->line);		\
-		}							\
-	}
-
-static void *my_shout_init_driver(struct audio_output *audio_output,
-				  const struct audio_format *audio_format,
-				  ConfigParam *param)
-{
-	struct shout_data *sd;
-	char *test;
-	int port;
-	char *host;
-	char *mount;
-	char *passwd;
-	const char *encoding;
-	unsigned protocol;
-	const char *user;
-	char *name;
-	BlockParam *block_param;
-	int public;
-
-	sd = new_shout_data();
-	sd->audio_output = audio_output;
-
-	if (shout_init_count == 0)
-		shout_init();
-
-	shout_init_count++;
-
-	check_block_param("host");
-	host = block_param->value;
-
-	check_block_param("mount");
-	mount = block_param->value;
-
-	check_block_param("port");
-
-	port = strtol(block_param->value, &test, 10);
-
-	if (*test != '\0' || port <= 0) {
-		FATAL("shout port \"%s\" is not a positive integer, line %i\n",
-		      block_param->value, block_param->line);
-	}
-
-	check_block_param("password");
-	passwd = block_param->value;
-
-	check_block_param("name");
-	name = block_param->value;
-
-	public = getBoolBlockParam(param, "public", 1);
-	if (public == CONF_BOOL_UNSET)
-		public = 0;
-
-	block_param = getBlockParam(param, "user");
-	if (block_param)
-		user = block_param->value;
-	else
-		user = "source";
-
-	block_param = getBlockParam(param, "quality");
-
-	if (block_param) {
-		int line = block_param->line;
-
-		sd->quality = strtod(block_param->value, &test);
-
-		if (*test != '\0' || sd->quality < -1.0 || sd->quality > 10.0) {
-			FATAL("shout quality \"%s\" is not a number in the "
-			      "range -1 to 10, line %i\n", block_param->value,
-			      block_param->line);
-		}
-
-		block_param = getBlockParam(param, "bitrate");
-
-		if (block_param) {
-			FATAL("quality (line %i) and bitrate (line %i) are "
-			      "both defined for shout output\n", line,
-			      block_param->line);
-		}
-	} else {
-		block_param = getBlockParam(param, "bitrate");
-
-		if (!block_param) {
-			FATAL("neither bitrate nor quality defined for shout "
-			      "output at line %i\n", param->line);
-		}
-
-		sd->bitrate = strtol(block_param->value, &test, 10);
-
-		if (*test != '\0' || sd->bitrate <= 0) {
-			FATAL("bitrate at line %i should be a positive integer "
-			      "\n", block_param->line);
-		}
-	}
-
-	check_block_param("format");
-
-	assert(audio_format != NULL);
-	sd->audio_format = *audio_format;
-
-	block_param = getBlockParam(param, "encoding");
-	if (block_param) {
-		if (0 == strcmp(block_param->value, "mp3"))
-			encoding = block_param->value;
-		else if (0 == strcmp(block_param->value, "ogg"))
-			encoding = block_param->value;
-		else
-			FATAL("shout encoding \"%s\" is not \"ogg\" or "
-			      "\"mp3\", line %i\n", block_param->value,
-			      block_param->line);
-	} else {
-		encoding = "ogg";
-	}
-
-	sd->encoder = shout_encoder_plugin_get(encoding);
-	if (sd->encoder == NULL)
-		FATAL("couldn't find shout encoder plugin for \"%s\" "
-		      "at line %i\n", encoding, block_param->line);
-
-	check_block_param("protocol");
-
-	block_param = getBlockParam(param, "protocol");
-	if (block_param) {
-		if (0 == strcmp(block_param->value, "shoutcast") &&
-		    0 != strcmp(encoding, "mp3"))
-			FATAL("you cannot stream \"%s\" to shoutcast, use mp3\n",
-			      encoding);
-		else if (0 == strcmp(block_param->value, "shoutcast"))
-			protocol = SHOUT_PROTOCOL_ICY;
-		else if (0 == strcmp(block_param->value, "icecast1"))
-			protocol = SHOUT_PROTOCOL_XAUDIOCAST;
-		else if (0 == strcmp(block_param->value, "icecast2"))
-			protocol = SHOUT_PROTOCOL_HTTP;
-		else
-			FATAL("shout protocol \"%s\" is not \"shoutcast\" or "
-			      "\"icecast1\"or "
-			      "\"icecast2\", line %i\n", block_param->value,
-			      block_param->line);
-	} else {
-		protocol = SHOUT_PROTOCOL_HTTP;
-	}
-
-	if (shout_set_host(sd->shout_conn, host) != SHOUTERR_SUCCESS ||
-	    shout_set_port(sd->shout_conn, port) != SHOUTERR_SUCCESS ||
-	    shout_set_password(sd->shout_conn, passwd) != SHOUTERR_SUCCESS ||
-	    shout_set_mount(sd->shout_conn, mount) != SHOUTERR_SUCCESS ||
-	    shout_set_name(sd->shout_conn, name) != SHOUTERR_SUCCESS ||
-	    shout_set_user(sd->shout_conn, user) != SHOUTERR_SUCCESS ||
-	    shout_set_public(sd->shout_conn, public) != SHOUTERR_SUCCESS ||
-	    shout_set_nonblocking(sd->shout_conn, 1) != SHOUTERR_SUCCESS ||
-	    shout_set_format(sd->shout_conn, sd->encoder->shout_format)
-	    != SHOUTERR_SUCCESS ||
-	    shout_set_protocol(sd->shout_conn, protocol) != SHOUTERR_SUCCESS ||
-	    shout_set_agent(sd->shout_conn, "MPD") != SHOUTERR_SUCCESS) {
-		FATAL("error configuring shout defined at line %i: %s\n",
-		      param->line, shout_get_error(sd->shout_conn));
-	}
-
-	/* optional paramters */
-	block_param = getBlockParam(param, "timeout");
-	if (block_param) {
-		sd->timeout = (int)strtol(block_param->value, &test, 10);
-		if (*test != '\0' || sd->timeout <= 0) {
-			FATAL("shout timeout is not a positive integer, "
-			      "line %i\n", block_param->line);
-		}
-	}
-
-	block_param = getBlockParam(param, "genre");
-	if (block_param && shout_set_genre(sd->shout_conn, block_param->value)) {
-		FATAL("error configuring shout defined at line %i: %s\n",
-		      param->line, shout_get_error(sd->shout_conn));
-	}
-
-	block_param = getBlockParam(param, "description");
-	if (block_param && shout_set_description(sd->shout_conn,
-						 block_param->value)) {
-		FATAL("error configuring shout defined at line %i: %s\n",
-		      param->line, shout_get_error(sd->shout_conn));
-	}
-
-	{
-		char temp[11];
-		memset(temp, 0, sizeof(temp));
-
-		snprintf(temp, sizeof(temp), "%u", sd->audio_format.channels);
-		shout_set_audio_info(sd->shout_conn, SHOUT_AI_CHANNELS, temp);
-
-		snprintf(temp, sizeof(temp), "%u", sd->audio_format.sample_rate);
-
-		shout_set_audio_info(sd->shout_conn, SHOUT_AI_SAMPLERATE, temp);
-
-		if (sd->quality >= -1.0) {
-			snprintf(temp, sizeof(temp), "%2.2f", sd->quality);
-			shout_set_audio_info(sd->shout_conn, SHOUT_AI_QUALITY,
-					     temp);
-		} else {
-			snprintf(temp, sizeof(temp), "%d", sd->bitrate);
-			shout_set_audio_info(sd->shout_conn, SHOUT_AI_BITRATE,
-					     temp);
-		}
-	}
-
-	if (sd->encoder->init_func(sd) != 0)
-		FATAL("shout: encoder plugin '%s' failed to initialize\n",
-		      sd->encoder->name);
-
-	return sd;
-}
-
-static int handle_shout_error(struct shout_data *sd, int err)
-{
-	switch (err) {
-	case SHOUTERR_SUCCESS:
-		break;
-	case SHOUTERR_UNCONNECTED:
-	case SHOUTERR_SOCKET:
-		ERROR("Lost shout connection to %s:%i: %s\n",
-		      shout_get_host(sd->shout_conn),
-		      shout_get_port(sd->shout_conn),
-		      shout_get_error(sd->shout_conn));
-		sd->shout_error = 1;
-		return -1;
-	default:
-		ERROR("shout: connection to %s:%i error: %s\n",
-		      shout_get_host(sd->shout_conn),
-		      shout_get_port(sd->shout_conn),
-		      shout_get_error(sd->shout_conn));
-		sd->shout_error = 1;
-		return -1;
-	}
-
-	return 0;
-}
-
-static int write_page(struct shout_data *sd)
-{
-	int err;
-
-	if (sd->buf.len == 0)
-		return 0;
-
-	shout_sync(sd->shout_conn);
-	err = shout_send(sd->shout_conn, sd->buf.data, sd->buf.len);
-	if (handle_shout_error(sd, err) < 0)
-		return -1;
-	sd->buf.len = 0;
-
-	return 0;
-}
-
-static void close_shout_conn(struct shout_data * sd)
-{
-	if (sd->opened) {
-		if (sd->encoder->clear_encoder_func(sd))
-			write_page(sd);
-	}
-
-	if (shout_get_connected(sd->shout_conn) != SHOUTERR_UNCONNECTED &&
-	    shout_close(sd->shout_conn) != SHOUTERR_SUCCESS) {
-		ERROR("problem closing connection to shout server: %s\n",
-		      shout_get_error(sd->shout_conn));
-	}
-
-	sd->opened = 0;
-}
-
-static void my_shout_finish_driver(void *data)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-
-	close_shout_conn(sd);
-
-	sd->encoder->finish_func(sd);
-	free_shout_data(sd);
-
-	shout_init_count--;
-
-	if (shout_init_count == 0)
-		shout_shutdown();
-}
-
-static void my_shout_drop_buffered_audio(void *data)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-	timer_reset(sd->timer);
-
-	/* needs to be implemented for shout */
-}
-
-static void my_shout_close_device(void *data)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-
-	close_shout_conn(sd);
-
-	if (sd->timer) {
-		timer_free(sd->timer);
-		sd->timer = NULL;
-	}
-}
-
-static int shout_connect(struct shout_data *sd)
-{
-	time_t t = time(NULL);
-	int state = shout_get_connected(sd->shout_conn);
-
-	/* already connected */
-	if (state == SHOUTERR_CONNECTED)
-		return 0;
-
-	/* waiting to connect */
-	if (state == SHOUTERR_BUSY && sd->conn_attempts != 0) {
-		/* timeout waiting to connect */
-		if ((t - sd->last_attempt) > sd->timeout) {
-			ERROR("timeout connecting to shout server %s:%i "
-			      "(attempt %i)\n",
-			      shout_get_host(sd->shout_conn),
-			      shout_get_port(sd->shout_conn),
-			      sd->conn_attempts);
-			return -1;
-		}
-
-		return 1;
-	}
-
-	/* we're in some funky state, so just reset it to unconnected */
-	if (state != SHOUTERR_UNCONNECTED)
-		shout_close(sd->shout_conn);
-
-	/* throttle new connection attempts */
-	if (sd->conn_attempts != 0 &&
-	    (t - sd->last_attempt) <= CONN_ATTEMPT_INTERVAL) {
-		return -1;
-	}
-
-	/* initiate a new connection */
-
-	sd->conn_attempts++;
-	sd->last_attempt = t;
-
-	state = shout_open(sd->shout_conn);
-	switch (state) {
-	case SHOUTERR_SUCCESS:
-	case SHOUTERR_CONNECTED:
-		return 0;
-	case SHOUTERR_BUSY:
-		return 1;
-	default:
-		ERROR("problem opening connection to shout server %s:%i "
-		      "(attempt %i): %s\n",
-		      shout_get_host(sd->shout_conn),
-		      shout_get_port(sd->shout_conn),
-		      sd->conn_attempts, shout_get_error(sd->shout_conn));
-		return -1;
-	}
-}
-
-static int open_shout_conn(void *data)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-	int status;
-
-	status = shout_connect(sd);
-	if (status != 0)
-		return status;
-
-	if (sd->encoder->init_encoder_func(sd) < 0) {
-		shout_close(sd->shout_conn);
-		return -1;
-	}
-
-	write_page(sd);
-
-	sd->shout_error = 0;
-	sd->opened = 1;
-	sd->tag_to_send = 1;
-	sd->conn_attempts = 0;
-
-	return 0;
-}
-
-static int my_shout_open_device(void *data,
-				struct audio_format *audio_format)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-
-	if (!sd->opened && open_shout_conn(sd) < 0)
-		return -1;
-
-	if (sd->timer)
-		timer_free(sd->timer);
-
-	sd->timer = timer_new(audio_format);
-
-	return 0;
-}
-
-static void send_metadata(struct shout_data * sd)
-{
-	static const int size = 1024;
-	char song[size];
-
-	if (!sd->opened || !sd->tag)
-		return;
-
-	if (sd->encoder->send_metadata_func(sd, song, size)) {
-		shout_metadata_add(sd->shout_meta, "song", song);
-		if (SHOUTERR_SUCCESS != shout_set_metadata(sd->shout_conn,
-							   sd->shout_meta)) {
-			ERROR("error setting shout metadata\n");
-			return;
-		}
-	}
-
-	sd->tag_to_send = 0;
-}
-
-static int my_shout_play(void *data,
-			 const char *chunk, size_t size)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-	int status;
-
-	if (!sd->timer->started)
-		timer_start(sd->timer);
-
-	timer_add(sd->timer, size);
-
-	if (sd->opened && sd->tag_to_send)
-		send_metadata(sd);
-
-	if (!sd->opened) {
-		status = open_shout_conn(sd);
-		if (status < 0) {
-			my_shout_close_device(sd);
-			return -1;
-		} else if (status > 0) {
-			timer_sync(sd->timer);
-			return 0;
-		}
-	}
-
-	if (sd->encoder->encode_func(sd, chunk, size)) {
-		my_shout_close_device(sd);
-		return -1;
-	}
-
-	if (write_page(sd) < 0) {
-		my_shout_close_device(sd);
-		return -1;
-	}
-
-	return 0;
-}
-
-static void my_shout_pause(void *data)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-	static const char silence[1020];
-	int ret;
-
-	/* play silence until the player thread sends us a command */
-
-	while (sd->opened && !audio_output_is_pending(sd->audio_output)) {
-		ret = my_shout_play(data, silence, sizeof(silence));
-		if (ret != 0)
-			break;
-	}
-}
-
-static void my_shout_set_tag(void *data,
-			     const struct tag *tag)
-{
-	struct shout_data *sd = (struct shout_data *)data;
-
-	if (sd->tag)
-		tag_free(sd->tag);
-	sd->tag = NULL;
-	sd->tag_to_send = 0;
-
-	if (!tag)
-		return;
-
-	sd->tag = tag_dup(tag);
-	sd->tag_to_send = 1;
-}
-
-const struct audio_output_plugin shoutPlugin = {
-	.name = "shout",
-	.init = my_shout_init_driver,
-	.finish = my_shout_finish_driver,
-	.open = my_shout_open_device,
-	.play = my_shout_play,
-	.pause = my_shout_pause,
-	.cancel = my_shout_drop_buffered_audio,
-	.close = my_shout_close_device,
-	.send_tag = my_shout_set_tag,
-};
-
-#else
-
-DISABLED_AUDIO_OUTPUT_PLUGIN(shoutPlugin)
-#endif
diff --git a/src/audioOutputs/audioOutput_shout.h b/src/audioOutputs/audioOutput_shout.h
deleted file mode 100644
index 2cfe68f29..000000000
--- a/src/audioOutputs/audioOutput_shout.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#ifndef AUDIO_OUTPUT_SHOUT_H
-#define AUDIO_OUTPUT_SHOUT_H
-
-#include "../output_api.h"
-
-#ifdef HAVE_SHOUT
-
-#include "../conf.h"
-#include "../timer.h"
-
-#include <shout/shout.h>
-
-struct shout_data;
-
-struct shout_encoder_plugin {
-	const char *name;
-	unsigned int shout_format;
-
-	int (*clear_encoder_func)(struct shout_data *sd);
-	int (*encode_func)(struct shout_data *sd,
-			   const char *chunk, size_t len);
-	void (*finish_func)(struct shout_data *sd);
-	int (*init_func)(struct shout_data *sd);
-	int (*init_encoder_func) (struct shout_data *sd);
-	/* Called when there is a new MpdTag to encode into the
-	   stream.  If this function returns non-zero, then the
-	   resulting song will be passed to the shout server as
-	   metadata.  This allows the Ogg encoder to send metadata via
-	   Vorbis comments in the stream, while an MP3 encoder can use
-	   the Shout Server's metadata API. */
-	int (*send_metadata_func)(struct shout_data *sd,
-				  char *song, size_t size);
-};
-
-struct shout_buffer {
-	unsigned char data[8192];
-	size_t len;
-};
-
-struct shout_data {
-	struct audio_output *audio_output;
-
-	shout_t *shout_conn;
-	shout_metadata_t *shout_meta;
-	int shout_error;
-
-	const struct shout_encoder_plugin *encoder;
-	void *encoder_data;
-
-	float quality;
-	int bitrate;
-
-	int opened;
-
-	struct tag *tag;
-	int tag_to_send;
-
-	int timeout;
-	int conn_attempts;
-	time_t last_attempt;
-
-	Timer *timer;
-
-	/* the configured audio format */
-	struct audio_format audio_format;
-
-	struct shout_buffer buf;
-};
-
-extern const struct shout_encoder_plugin shout_mp3_encoder;
-extern const struct shout_encoder_plugin shout_ogg_encoder;
-
-#endif
-
-#endif
diff --git a/src/audioOutputs/audioOutput_shout_mp3.c b/src/audioOutputs/audioOutput_shout_mp3.c
deleted file mode 100644
index 722079b29..000000000
--- a/src/audioOutputs/audioOutput_shout_mp3.c
+++ /dev/null
@@ -1,189 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "../output_api.h"
-
-#ifdef HAVE_SHOUT_MP3
-
-#include "../utils.h"
-#include "audioOutput_shout.h"
-#include <lame/lame.h>
-
-struct lame_data {
-	lame_global_flags *gfp;
-};
-
-
-static int shout_mp3_encoder_init(struct shout_data *sd)
-{
-	struct lame_data *ld;
-
-	if (NULL == (ld = xmalloc(sizeof(*ld))))
-		FATAL("error initializing lame encoder data\n");
-	sd->encoder_data = ld;
-
-	return 0;
-}
-
-static int shout_mp3_encoder_clear_encoder(struct shout_data *sd)
-{
-	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
-	struct shout_buffer *buf = &sd->buf;
-	int ret;
-
-	if ((ret = lame_encode_flush(ld->gfp, buf->data + buf->len,
-				     buf->len)) < 0)
-		ERROR("error flushing lame buffers\n");
-
-	return (ret > 0);
-}
-
-static void shout_mp3_encoder_finish(struct shout_data *sd)
-{
-	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
-
-	lame_close(ld->gfp);
-	ld->gfp = NULL;
-}
-
-static int shout_mp3_encoder_init_encoder(struct shout_data *sd)
-{
-	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
-
-	if (NULL == (ld->gfp = lame_init())) {
-		ERROR("error initializing lame encoder for shout\n");
-		return -1;
-	}
-
-	if (sd->quality >= -1.0) {
-		if (0 != lame_set_VBR(ld->gfp, vbr_rh)) {
-			ERROR("error setting lame VBR mode\n");
-			return -1;
-		}
-		if (0 != lame_set_VBR_q(ld->gfp, sd->quality)) {
-			ERROR("error setting lame VBR quality\n");
-			return -1;
-		}
-	} else {
-		if (0 != lame_set_brate(ld->gfp, sd->bitrate)) {
-			ERROR("error setting lame bitrate\n");
-			return -1;
-		}
-	}
-
-	if (0 != lame_set_num_channels(ld->gfp,
-				       sd->audio_format.channels)) {
-		ERROR("error setting lame num channels\n");
-		return -1;
-	}
-
-	if (0 != lame_set_in_samplerate(ld->gfp,
-					sd->audio_format.sample_rate)) {
-		ERROR("error setting lame sample rate\n");
-		return -1;
-	}
-
-	if (0 > lame_init_params(ld->gfp))
-		FATAL("error initializing lame params\n");
-
-	return 0;
-}
-
-static int shout_mp3_encoder_send_metadata(struct shout_data *sd,
-					   char * song, size_t size)
-{
-	char artist[size];
-	char title[size];
-	int i;
-	struct tag *tag = sd->tag;
-
-	strncpy(artist, "", size);
-	strncpy(title, "", size);
-
-	for (i = 0; i < tag->numOfItems; i++) {
-		switch (tag->items[i]->type) {
-		case TAG_ITEM_ARTIST:
-			strncpy(artist, tag->items[i]->value, size);
-			break;
-		case TAG_ITEM_TITLE:
-			strncpy(title, tag->items[i]->value, size);
-			break;
-
-		default:
-			break;
-		}
-	}
-	snprintf(song, size, "%s - %s", title, artist);
-
-	return 1;
-}
-
-static int shout_mp3_encoder_encode(struct shout_data *sd,
-				    const char * chunk, size_t len)
-{
-	unsigned int i;
-	int j;
-	float (*lamebuf)[2];
-	struct shout_buffer *buf = &(sd->buf);
-	unsigned int samples;
-	int bytes = audio_format_sample_size(&sd->audio_format);
-	struct lame_data *ld = (struct lame_data *)sd->encoder_data;
-	int bytes_out;
-
-	samples = len / (bytes * sd->audio_format.channels);
-	/* rough estimate, from lame.h */
-	lamebuf = xmalloc(sizeof(float) * (1.25 * samples + 7200));
-
-	/* this is for only 16-bit audio */
-
-	for (i = 0; i < samples; i++) {
-		for (j = 0; j < sd->audio_format.channels; j++) {
-			lamebuf[j][i] = *((const int16_t *) chunk);
-			chunk += bytes;
-		}
-	}
-
-	bytes_out = lame_encode_buffer_float(ld->gfp, lamebuf[0], lamebuf[1],
-					     samples, buf->data,
-					     sizeof(buf->data) - buf->len);
-	free(lamebuf);
-
-	if (0 > bytes_out) {
-		ERROR("error encoding lame buffer for shout\n");
-		lame_close(ld->gfp);
-		ld->gfp = NULL;
-		return -1;
-	} else
-		buf->len = bytes_out; /* signed to unsigned conversion */
-
-	return 0;
-}
-
-const struct shout_encoder_plugin shout_mp3_encoder = {
-	"mp3",
-	SHOUT_FORMAT_MP3,
-
-	shout_mp3_encoder_clear_encoder,
-	shout_mp3_encoder_encode,
-	shout_mp3_encoder_finish,
-	shout_mp3_encoder_init,
-	shout_mp3_encoder_init_encoder,
-	shout_mp3_encoder_send_metadata,
-};
-
-#endif
diff --git a/src/audioOutputs/audioOutput_shout_ogg.c b/src/audioOutputs/audioOutput_shout_ogg.c
deleted file mode 100644
index 5983b4d89..000000000
--- a/src/audioOutputs/audioOutput_shout_ogg.c
+++ /dev/null
@@ -1,306 +0,0 @@
-/* the Music Player Daemon (MPD)
- * Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
- * This project's homepage is: http://www.musicpd.org
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
- * GNU General Public License for more details.
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
- */
-
-#include "audioOutput_shout.h"
-
-#ifdef HAVE_SHOUT_OGG
-
-#include "../utils.h"
-#include <vorbis/vorbisenc.h>
-
-struct ogg_vorbis_data {
-	ogg_stream_state os;
-	ogg_page og;
-	ogg_packet op;
-	ogg_packet header_main;
-	ogg_packet header_comments;
-	ogg_packet header_codebooks;
-
-	vorbis_dsp_state vd;
-	vorbis_block vb;
-	vorbis_info vi;
-	vorbis_comment vc;
-};
-
-static void add_tag(struct ogg_vorbis_data *od, const char *name, char *value)
-{
-	if (value) {
-		union {
-			const char *in;
-			char *out;
-		} u = { .in = name };
-		vorbis_comment_add_tag(&od->vc, u.out, value);
-	}
-}
-
-static void copy_tag_to_vorbis_comment(struct shout_data *sd)
-{
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
-	if (sd->tag) {
-		int i;
-
-		for (i = 0; i < sd->tag->numOfItems; i++) {
-			switch (sd->tag->items[i]->type) {
-			case TAG_ITEM_ARTIST:
-				add_tag(od, "ARTIST", sd->tag->items[i]->value);
-				break;
-			case TAG_ITEM_ALBUM:
-				add_tag(od, "ALBUM", sd->tag->items[i]->value);
-				break;
-			case TAG_ITEM_TITLE:
-				add_tag(od, "TITLE", sd->tag->items[i]->value);
-				break;
-
-			default:
-				break;
-			}
-		}
-	}
-}
-
-static int copy_ogg_buffer_to_shout_buffer(ogg_page *og,
-					   struct shout_buffer *buf)
-{
-	if (sizeof(buf->data) - buf->len >= (size_t)og->header_len) {
-		memcpy(buf->data + buf->len,
-		       og->header, og->header_len);
-		buf->len += og->header_len;
-	} else {
-		ERROR("%s: not enough buffer space!\n", __func__);
-		return -1;
-	}
-
-	if (sizeof(buf->data) - buf->len >= (size_t)og->body_len) {
-		memcpy(buf->data + buf->len,
-		       og->body, og->body_len);
-		buf->len += og->body_len;
-	} else {
-		ERROR("%s: not enough buffer space!\n", __func__);
-		return -1;
-	}
-
-	return 0;
-}
-
-static int flush_ogg_buffer(struct shout_data *sd)
-{
-	struct shout_buffer *buf = &sd->buf;
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-	int ret = 0;
-
-	if (ogg_stream_flush(&od->os, &od->og))
-		ret = copy_ogg_buffer_to_shout_buffer(&od->og, buf);
-
-	return ret;
-}
-
-static int send_ogg_vorbis_header(struct shout_data *sd)
-{
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
-	vorbis_analysis_headerout(&od->vd, &od->vc,
-				  &od->header_main,
-				  &od->header_comments,
-				  &od->header_codebooks);
-
-	ogg_stream_packetin(&od->os, &od->header_main);
-	ogg_stream_packetin(&od->os, &od->header_comments);
-	ogg_stream_packetin(&od->os, &od->header_codebooks);
-
-	return flush_ogg_buffer(sd);
-}
-
-static void finish_encoder(struct ogg_vorbis_data *od)
-{
-	vorbis_analysis_wrote(&od->vd, 0);
-
-	while (vorbis_analysis_blockout(&od->vd, &od->vb) == 1) {
-		vorbis_analysis(&od->vb, NULL);
-		vorbis_bitrate_addblock(&od->vb);
-		while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
-			ogg_stream_packetin(&od->os, &od->op);
-		}
-	}
-}
-
-static int shout_ogg_encoder_clear_encoder(struct shout_data *sd)
-{
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-	int ret;
-
-	finish_encoder(od);
-	if ((ret = ogg_stream_pageout(&od->os, &od->og)))
-		copy_ogg_buffer_to_shout_buffer(&od->og, &sd->buf);
-
-	vorbis_comment_clear(&od->vc);
-	ogg_stream_clear(&od->os);
-	vorbis_block_clear(&od->vb);
-	vorbis_dsp_clear(&od->vd);
-	vorbis_info_clear(&od->vi);
-
-	return ret;
-}
-
-static void shout_ogg_encoder_finish(struct shout_data *sd)
-{
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
-	if (od) {
-		free(od);
-		sd->encoder_data = NULL;
-	}
-}
-
-static int shout_ogg_encoder_init(struct shout_data *sd)
-{
-	struct ogg_vorbis_data *od;
-
-	if (NULL == (od = xmalloc(sizeof(*od))))
-		FATAL("error initializing ogg vorbis encoder data\n");
-	sd->encoder_data = od;
-
-	return 0;
-}
-
-static int reinit_encoder(struct shout_data *sd)
-{
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
-	vorbis_info_init(&od->vi);
-
-	if (sd->quality >= -1.0) {
-		if (0 != vorbis_encode_init_vbr(&od->vi,
-						sd->audio_format.channels,
-						sd->audio_format.sample_rate,
-						sd->quality * 0.1)) {
-			ERROR("error initializing vorbis vbr\n");
-			vorbis_info_clear(&od->vi);
-			return -1;
-		}
-	} else {
-		if (0 != vorbis_encode_init(&od->vi,
-					    sd->audio_format.channels,
-					    sd->audio_format.sample_rate, -1.0,
-					    sd->bitrate * 1000, -1.0)) {
-			ERROR("error initializing vorbis encoder\n");
-			vorbis_info_clear(&od->vi);
-			return -1;
-		}
-	}
-
-	vorbis_analysis_init(&od->vd, &od->vi);
-	vorbis_block_init(&od->vd, &od->vb);
-	ogg_stream_init(&od->os, rand());
-	vorbis_comment_init(&od->vc);
-
-	return 0;
-}
-
-static int shout_ogg_encoder_init_encoder(struct shout_data *sd)
-{
-	if (reinit_encoder(sd))
-		return -1;
-
-	if (send_ogg_vorbis_header(sd)) {
-		ERROR("error sending ogg vorbis header for shout\n");
-		return -1;
-	}
-
-	return 0;
-}
-
-static int shout_ogg_encoder_send_metadata(struct shout_data *sd,
-					   mpd_unused char * song,
-					   mpd_unused size_t size)
-{
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
-	shout_ogg_encoder_clear_encoder(sd);
-	if (reinit_encoder(sd))
-		return 0;
-
-	copy_tag_to_vorbis_comment(sd);
-
-	vorbis_analysis_headerout(&od->vd, &od->vc,
-				  &od->header_main,
-				  &od->header_comments,
-				  &od->header_codebooks);
-
-	ogg_stream_packetin(&od->os, &od->header_main);
-	ogg_stream_packetin(&od->os, &od->header_comments);
-	ogg_stream_packetin(&od->os, &od->header_codebooks);
-
-	flush_ogg_buffer(sd);
-
-	return 0;
-}
-
-static int shout_ogg_encoder_encode(struct shout_data *sd,
-				    const char *chunk, size_t size)
-{
-	struct shout_buffer *buf = &sd->buf;
-	unsigned int i;
-	int j;
-	float **vorbbuf;
-	unsigned int samples;
-	int bytes = audio_format_sample_size(&sd->audio_format);
-	struct ogg_vorbis_data *od = (struct ogg_vorbis_data *)sd->encoder_data;
-
-	samples = size / (bytes * sd->audio_format.channels);
-	vorbbuf = vorbis_analysis_buffer(&od->vd, samples);
-
-	/* this is for only 16-bit audio */
-
-	for (i = 0; i < samples; i++) {
-		for (j = 0; j < sd->audio_format.channels; j++) {
-			vorbbuf[j][i] = (*((const int16_t *) chunk)) / 32768.0;
-			chunk += bytes;
-		}
-	}
-
-	vorbis_analysis_wrote(&od->vd, samples);
-
-	while (1 == vorbis_analysis_blockout(&od->vd, &od->vb)) {
-		vorbis_analysis(&od->vb, NULL);
-		vorbis_bitrate_addblock(&od->vb);
-
-		while (vorbis_bitrate_flushpacket(&od->vd, &od->op)) {
-			ogg_stream_packetin(&od->os, &od->op);
-		}
-	}
-
-	if (ogg_stream_pageout(&od->os, &od->og))
-		copy_ogg_buffer_to_shout_buffer(&od->og, buf);
-
-	return 0;
-}
-
-const struct shout_encoder_plugin shout_ogg_encoder = {
-	"ogg",
-	SHOUT_FORMAT_VORBIS,
-
-	shout_ogg_encoder_clear_encoder,
-	shout_ogg_encoder_encode,
-	shout_ogg_encoder_finish,
-	shout_ogg_encoder_init,
-	shout_ogg_encoder_init_encoder,
-	shout_ogg_encoder_send_metadata,
-};
-
-#endif